Home

PortaBilling: User Manual

Contents

1. 57 User Authentication 57 Call PARKING isisisi 59 Paging Intercom Calls 61 SIP 0000 01110000 62 Direct Incoming Calls to B2BUA 62 VOIP from Vendor Connection 63 Legal Call Intercept 64 Secure Callli th hisesssciscsssessinsstasacosozsstsasbascdoesusa 65 Voice VPN Rating 65 Voice Qn n t 1881110 0 unes 65 Support for Caller Identity and Privacy Flags 66 3 IP Centrex Features 68 HOW TO 222214641666466 408430430 74 configure my Cisco gateway to accept incoming SIP calls and terminate them to a telephony network 75 configure my Cisco gateway to send outgoing calls using SIP 76 configure my Cisco gateway for PSTN gt SIP service oo 77 2000 2009 PortaOne Inc All rights Reserved www portaone com 3 Porta SIP 2000 2009 PortaOne I nc All rights Reserved www portaone com System Concepts Support incoming H323 and SIP calls on the same gateway 77 configure my Cisco ATA186 to work with PortaSIP 78 provide services to and bill a customer who has a SIP enabled gateway but no authoriza
2. 2000 2009 PortaOne Inc All rights Reserved www portaone com 69 Porta SIP System Concepts location and then resume the call from any other station in the Centrex by dialing a pickup code Supported by PortaSwitch in order to use this feature the customer should define a call parking prefix in his call features configuration Then when a phone conversation is under way the user can simply place the call on hold and dial the specified call parking prefix The dynamically assigned retrieval code will be heard this can be dialed from any phone in the customer s IP Centrex group to retrieve the conversation i e connect the call to that phone It is also possible to quickly retrieve a call from the original phone by dialing a special de park code Call Restrictions Station Restrictions Feature description Prevents certain types of calls from being made or received by particular stations For example phones in public areas can be blocked from originating calls to external numbers so as to prevent unauthorized users from incurring toll charges Phones in certain areas may be blocked from receiving external calls in order to limit employees ability to take personal calis A wide variety of restrictions are available covering incoming calls outgoing calis toll restrictions code restrictions and differential treatment for internal and external calls Provided using the tariff configuration in PortaBilling
3. Call Return Feature description Allows the user to originate a call to the last party or number that called the user regardless of whether the user answered the original call or knows the caller s identity Provided by the IP phone dial the 69 code to use this feature Call Transfer Feature description Transfers an existing call to another party inside or outside the Centrex group Supported by PortaSwitch Call Waiting Feature description A feature that allows users to be alerted of one or more calls awaiting connection during a current conversation Users are normally notified by a short tone on the phone or by use of the caller ID feature Then they can answer the second call while the first one is still on hold Control Call Waiting Feature description Enables disables delivery of the call waiting feature to IP phones allowing administrators to control call waiting for a specific account This ensures that the feature is supplied only to users who have it activated on the PortaS witch side regardless of whether it is enabled on the IP phone itself Supported by PortaSwitch 2000 2009 PortaOne Inc All rights Reserved www portaone com 70 Porta SIP System Concepts Caller ID Feature description Allows the user to identify the name and telephone number of a calling party before answering an incoming call Supported by PortaSwitch the phone must have a display to show the caller ID Caller ID on Cal
4. lt 1000 gt xxxx Call Forwarding on Busy Feature description Automatically routes incoming calls for a given extension to another pre selected number when the first extension is busy This feature is implemented by provisioning the follow me service choose Follow me when unavailable and activating the cfwd Busy Serv supplementary service on the IP phone Use the 90 code to activate this feature and 91 to deactivate it Call Forwarding on Don t Answer Feature description Automatically routes incoming calls for a given extension to another pre selected number when there is no answer after a specified number of rings This feature is implemented by provisioning the follow me service choose Follow me when unavailable then set the ring timeout parameter in follow me You may also utilize this feature on the IP phone itself by activating the cfwd No Ans Serv supplementary service Use the 92 code to activate this feature and 93 to deactivate it Call Forwarding to Multiple Simultaneous Extensions Feature description Indicates the number of forwarded calls originally dialed to the same Centrex extension which may occur simultaneously This feature may be implemented similarly to other call forwarding scenarios only this time the follow me service should be provisioned with a simultaneous ring option Call Park Call Pickup Feature description Allows the user to place a call on hold move to a different
5. 2 e Based on the results the PortaSIP presence server sends a notification response 3 via the PortaSIP proxy server 4 back to the user agent e The user agent presentity sends a PUBLISH request to the PortaSIP proxy server 5 if authorized successfully the request is forwarded to the PortaSIP presence server 6 e The presence server sends a NOTIFY request to the PortaSIP proxy server 7 which identifies SIP user agents watchers subscribing to presence for the given user and forwards them a NOTIFY request 8 Instant Messaging Instant Messaging IM is defined as the exchange of text messages between two users in real time As a service IM is always coupled with the presence service see earlier in this chapter For example when a friend comes online a user can be notified of this and have the option of sending his friend an instant message Supported by wide range of multimedia clients such as MS Messenger instant messaging can be easily used to post messages from any computer or mobile device 2000 2009 PortaOne Inc All rights Reserved www portaone com 23 Porta SIP System Concepts Porta SIP Presence status Presence status SIP SIP Messaging Messaging Re Re PortaSIP includes an advanced messaging module that enables online messaging server side message storage for offline users so they can receive messages later and the option of maintaining full message history o
6. activities e g when an unsuspecting victim is transferred to a very expensive international destination 2000 2009 PortaOne Inc All rights Reserved www portaone com 46 Porta SIP System Concepts Unattended blind transfer Porta M Billing PSTN GW Phone C 9 SIP phone A SIP phone B e A dials B s phone number 1 e PortaSIP sends the incoming call to B 2 when B answers the call is established between A and B 3 e Ata certain moment in the conversation B performs a call transfer REFER to C 4 e PortaSIP intercepts this message and sends an authorization request to PortaBilling to check if B is allowed to send a call to this destination and to obtain the routing 5 In the case of a positive reply PortaSIP starts processing the call transfer e The call leg going to B is canceled 6 since B is no longer a participant in this call a new outgoing call is sent to C 7 and A the transferred party receives a re INVITE message 8 e Finally the call is established between A and C 9 e When either A or C hangs up the call is terminated and two accounting records are sent to the billing 10 one is for the A gt B call charged to its originator A and the other for the A gt C call likewise charged to its originator B Assuming that A spoke to B for 5 minutes before B initiated the transfer then A spoke to C for another 10 minutes the call charges CDRs will look like this e Unde
7. his first usage flag is reset and no further messages will be played User Authentication In general every incoming call to PortaSIP must be authorized in order to ensure that it comes from a legitimate customer of yours 2000 2009 PortaOne Inc All rights Reserved www portaone com 57 Porta SIP System Concepts Digest authorization PortaSIP UA ser bZbua ABA server 70 68 0 213 216 231 44 34 216 231 44 34 time Sipura SPAZOOO 3 1 5 PortaSIP PortaSIP PorteBilling 8 Dec l l l i 10 19 48 G gt A 101 1 INVITE gt l 10 19 48 lt A 101 I 100 trying i 10 19 48 gt A 101 I INVITE gt I 10 19 48 lt A 101 I 401 Unauthor 10 19 48 gt A 10l A ACK gt I 10 19 48 l 10 19 48 10 19 48 INVT l 10 19 48 lt A 102 1 100 trying 1 10 19 48 gt A 102 I INVITE gt 10 19 48 lt 102 1 100 Trying 10 19 48 I gt Authorization request gt 10 19 48 I lt Auth request accented fA When the first INVITE request arrives from a SIP phone the SIP server replies with 401 Unauthorized and provides the SIP UA with a challenge a long string of randomly generated characters The SIP UA must compute a response using this challenge a username a password and some other attributes with the MD5 algorithm This response is then sent back to the SIP server in another INVITE request The
8. 2009 PortaOne Inc All rights Reserved www portaone com 63 Porta SIP System Concepts Otherwise 1e if this call is indeed coming via a VoIP from vendor connection PortaBilling will compare the username and password supplied in the authorization request with those defined in the vendor account associated with this connection e If authentication succeeds 5 i e the call is indeed being sent by your vendor PortaBilling will apply the connection s translation rules and check whether the dialed number belongs to one of your accounts 1234 If it does not the call will be refused since there has probably been a configuration error so that the vendor is routing international traffic to your network e PortaSIP receives the routing information for the call 6 and so now recognizes that the call should be sent to one of your SIP phones 7 Follow me UM parameters and other related information are provided as well One very important point is that this call will be charged to the account which receives the call e After the call is disconnected the called account is charged for the call 8 and the costs of the call are calculated for the vendor Legal Call Intercept As an ITSP you may be requested to enable law enforcement agencies to monitor a certain subscriber s calls This may be required in accordance with the Communications Assistance for Law Enforcement Act of 1994 CALEA or some other law applicable in the country
9. Account phone line j ey pas IP phone inventory record MAC address gt file me lt 5 7 IP phone profile 1 General parame N br L i D 9000 000 as Request for provisioning information gt 5 Configuration fle 8 000 IP Phone The config file is specific to each user agent as it contains information such as username and password thus the user agent must retrieve its own designated config file The following are defined in the billing configuration e The IP phone profile so that the system knows which generic properties e g preferred codec to place in the configuration file e An entry about the specific IP phone in the IP phone inventory including the device s MAC address with a specific profile assigned to it e The IP phone or in the case of a multi line device a port on the phone is assigned to a specific account in the billing Auto provisioning will only work if your IP phone knows the address of A your provisioning server If you buy IP phones retail you will probably have to change the address of the provisioning server on every phone manually However if you place a large enough order with a specific vendor these settings can be pre configured by him so that you may deliver an IP phone directly to the end user without even unwrapping it IP Phone Inventory The IP phone directory allows you to keep track of IP devices SIP phon
10. Administration FAQ Troubleshooting Common Problems No or one way audio during SIP Phone SIP Phone calls This problem usually means that one or both phones are behind a NAT firewall Unfortunately unless the RTP Proxy is turned on or certain smart SIP phones NAT routers are used there is no way to guarantee proper performance in such cases see Nat Traversal section for details One way audio during SIP Phone Cisco gateway calls This problem can occur if the Cisco GW is not configured properly Please check that the GW contains the following in its IOS configuration sip ua nat symmetric check media src have problems when trying to use SIP phone X made by vendor Y with PortaSIP Unfortunately not all of the many SIP phones available on the market today fully comply with the SIP standard especially low end products We use Cisco ATA 186 as a reference phone and the Cisco ATA PortaSIP combination has been thoroughly tested If you are unable to get your third party vendor SIP phone working properly follow the instructions below e Make sure the phone has been configured properly with such parameters as account ID password SIP server address etc Consult the product documentation regarding other configuration settings e Check the PortaSIP and PortaBilling logs to ensure that there is not a problem with the account you are trying to use for example an expired or blocked account e Connect the Cisco ATA
11. Inc All rights Reserved www portaone com 97 Porta SIP Appendices 3 Click the Configure settings radio button and enter the Server name of IP address using either the IP address of the PortaSIP server or its name in DNS Make sure that the UDP radio button is selected then click OK SIP Communications Service Connection Configuration 4 Sign out and then sign in again You should see the pop up dialog below Fill it in as follows Sign in name in the form username address where username is the name of the appropriate account in PB and address is either the IP address of the PortaSIP server or its name in DNS Enter the name of the appropriate PB account as the User Name and the appropriate account password as the Password then click OK You should now see your status change to online Sign In to a SIP Communications Service 0118000 demo portaone com 0118000 2000 2009 PortaOne I nc All rights Reserved www portaone com 98 Porta SIP Appendices 5 To make a call click the Action item in the main menu then select Start Voice Conversation Click the Other tab making sure that Communications Service is selected in the drop down Service box and enter the phone number in the Enter e mail address field as shown below Finally click OK to place a call Start a Yoice Conversation x Enter the e mail address of the p
12. a specific city or county e An E911 provider is the company which delivers emergency calls to the PSAP Basically when a customer dials an emergency number he should be connected to the PSAP which is responsible for his location The PSAP must immediately obtain the customer s exact address e g including floor number so that if the customer is incapable of providing his address information an emergency response team may still reach him How is this done E911 service providers It is virtually impossible for an ITSP to establish a connection with every PSAP in a given country and meet all of their requirements basically for the same reason why it is impossible for an ITSP to establish a direct interconnection with every telco operator in a country Fortunately this is not necessary as there are companies who provide E911 services in a manner very similar to companies that offer wholesale call termination you send a call to their network and they deliver it to the designated destination Currently there are several companies in the US who provide these sort of services e g Intrado Dash911 and their number will probably increase Naturally local E911 providers will be found in other countries as well 2000 2009 PortaOne Inc All rights Reserved www portaone com 43 Porta SIP System Concepts To accommodate the demand for working with different providers PortaBilling uses a plugin model similar to that used for onli
13. and the server If no packets are sent in either direction over a certain period of time the NAT router regards the connection as terminated and removes this tunnel If an IP phone behind NAT sends data for this connection a new tunnel will be created and the functionality restored However if the SIP server tries to send data incoming call information after the NAT tunnel has been closed NAT will reject these packets since it has no information as to where they should be sent on LAN This may create problems because if a NAT router removes a tunnel too soon an IP phone may not receive some incoming calls To prevent this situation PortaSIP includes the NAThelper module which periodically sends small ping packets to registered SIP phones These packets are small and so do not create any significant network traffic but they are sent often enough so that the NAT router keeps the connection open 2000 2009 PortaOne Inc All rights Reserved www portaone com 56 Porta SIP System Concepts Keep alive Call Monitoring When a SIP phone goes offline during a phone conversation e g an Internet line is down the SIP server may not be aware of this fact So if the remote party does not hang up e g there is an automated IVR or a problem with disconnect supervision this call may stay in the active state for a long time To prevent this situation PortaSIP has a keep alive functionality e Cus
14. are joined together e Speakerphone mode is activated immediately on User B s phone SIP TAPI SIP TAPI is a TAPI driver that enables the SIP click2dial functionality for TAPI applications like MS Outlook a gis SIP phone A SIP phone B e A installs the SIP TAPI driver on his computer 0 e A clicks on the phone icon in his MS Outlook contact list to initiate a call 1 e The SIP TAPI client sends an INVITE to PortaSIP requesting a call to A s IP phone 2 and the IP phone starts ringing e A answers his phone 3 e The SIP TAPI client sends a call transfer message to A s phone requesting an outgoing call to B 4 e B answers his phone and A and B are connected 5 Direct Incoming Calls to B2BUA During the life of a VoIP call PortaSIP and the remote SIP UA exchange various SIP messages B2BUA is the originator or recipient of these messages but every message passes through the SIP proxy This is 2000 2009 PortaOne Inc All rights Reserved www portaone com 62 Porta SIP System Concepts necessary for several reasons the most important of them being the fact that the SIP proxy must perform NAT traversal However if a call arrives from a remote gateway or IP PBX running on a public IP address NAT traversal is not required and there is no need to engage the SIP proxy in the SIP message exchange In this case B2BUA may accept a direct incoming connection from a remote SIP UA on a public IP
15. be activated and configured for an individual customer and account only Also whereas only the administrator can manage a tariff plan call barring can be provisioned by end users themselves e g parents prohibiting calls to a dubious premium number on their child s phone or a small business owner blocking outgoing international calls on a public phone in his caf 2000 2009 PortaOne Inc All rights Reserved www portaone com 60 Porta SIP System Concepts When the Call Barring service feature is activated as part of normal call authorization the system checks whether a dialed number matches any pattern specified in the call barring classes If it does and if call barring has been activated for that class the call is rejected A call barring class covers a specific set of phone numbers that the customer should potentially be denied access to In this regard a call barring class is very similar to a destination group The difference is that while a destination group can only contain pre defined destination prefixes a call barring class operates with a mixture of patterns e g 448 any number starting with 448 and actual phone numbers e g 44810010099 This lets you fine tune call barring options without creating excessive destination prefixes Definition of the various call barring classes such as Mobiles International etc is done globally in the Call Barring tab under Company Info Barring of a specific
16. call information in such a way as to ensure the desired privacy Even if an end user requests that his identity be hidden from the called party some vendors still request that his identification information be sent to them so they can record this information for various purposes such as abuse prevention or law enforcement they will then take care of hiding it from the final recipient This actually means that PortaSwitch must send normal caller information along with a privacy flag that tells the vendor to withhold caller info from the final call recipient However many other vendors do not have the capability to process privacy flags properly In this case PortaSwitch must remove the Caller ID from the call information before sending the call to such a carrier s network Since a vendor s capabilities in this respect cannot be determined at the time a call is routed to his network the desired method should be selected in the vendor s connection configuration beforehand Then the proper method will be used whenever a call with a privacy request is sent to that particular carrier i 4 Edit Connection gt Save fe Save amp Close Close Load Objects gt I Logout 8 Log Description Demo 5 Type VoIP to Vendor Service Type Voice Calls Routing Criteria None E General Info Remote IP 193 28 87 120 RTP Proxying Direct Tariff vendor Mi C
17. com 195 70 140 2 LL PortaSIP instance Environm Eaa E d sip supercall net 5 e 195 70 140 3 gt PA Every virtual SIP server acts as an independent PortaSIP installation Customers of env 8 The virtual SIP instance resides in the var sipenv lt IP gt directory where lt IP gt is the IP address allocated to this SIP instance e g for a PortaSIP working on IP address 120 34 56 78 it will be var sipenv 120 34 56 78 Inside the sipenv directory there are several sub directories the most important ones being e etc this subdirectory contains a master configuration file for the SIP instance and config files for the individual modules e log PortaSIP log file sip log and copies of the log file for previous days are located here Since this latest release you can configure the network interface on the A SIP server using the web interface This may be done by using the SIP Environments page which is accessible to administrators For more information about creating and editing PortaSIP instances via the web see the section on STP Environments in the PortaBilling Web Reference Guide However since improperly configuring the network interface on the SIP server will render all of your SIP services useless you should not hesitate to contact the PortaOne support team for assistance with configuration 2000 2009 PortaOne Inc All rights Reserved www portaone com 25 Porta SIP S
18. e This call is routed through the telecom network to gateway GW NY 01 When the incoming call arrives on the gateway 1 it starts a special TCL application PSTN2SIP to handle this call This application does several things o Converts the phone number to the E 164 format so that 2027810003 become 12027810003 o Performs authorization in the billing 2 whether A is allowed to receive incoming telephony calls from GW NY 01 and if you charge for incoming calls what is the maximum call time allowed based on A s current balance 3 One important point is that authorization should happen without a password check since the application does not know the valid password for the SIP account Starts outgoing call to 12027810003 o Starts the timer once the call is established disconnecting the call when the maximum call duration is exceeded o The gateway is configured such that it knows that calls to 1202781 numbers should be sent to the PortaSIP server thus it sends an INVITE to PortaSIP 4 NOTE The gateway cannot make this call on behalf of A since even if we know A s account ID we do not know A s password therefore such a call will be rejected In addition Cisco gateways currently do not support INVITE with authorization 2000 2009 PortaOne Inc All rights Reserved www portaone com 18 Porta SIP System Concepts e PortaSIP receives the INVITE but without authorization information So the Port
19. is no such simple and universal NAT traversal solution There are 3 ways of dealing with this problem 1 Insert an RTP proxy integrated with the SIP Server into the RTP path The RTP proxy can then perform the same trick for the media stream as the SIP Server does for signaling identify the real source IP address UDP port for each party and use these addresses ports as targets for RTP rather than using the private addresses ports indicated by the UAs This method helps in all cases where properly configured UAs supporting symmetric media are used However it adds another hop in media propagation thus increasing audio delay and possibly decreasing quality due to greater packet loss 2 Assume that the NAT will not change the UDP port when resending an RTP stream from its WAN interface in which case the SIP Server can correct the IP address for the RTP stream in SIP messages This method is quite unreliable in some cases it works while in others it fails 3 Use smart UAs or NAT routers or a combination of both which are able to figure out the correct WAN IP address port for the media by themselves There are several technologies available for this purpose such as STUN UPnP and so on A detailed description of them lies beyond the scope of this document but may easily be found on the Internet Which NAT Traversal Method is the Best There is no ideal solution since all methods have their own advantages and drawbacks
20. main advantage of this method is that the actual password is never transferred over the Internet and there is no chance of recovering the password by monitoring challenge response pairs Such digest authentication provides a secute and flexible way to identify whether a remote SIP device is indeed a legitimate customer Authorization based on IP address Unfortunately some SIP UAs e g the Cisco AS5300 5350 gateway do not support digest authentication for outgoing calls This means that when the SIP UA receives a 401 Unauthorized reply from the SIP server it will simply drop the call as it is unable to proceed with call setup In this case PortaSIP may be configured to detect that a call is coming from a digest incapable SIP UA and perform authorization based on the SIP UA s remote IP The User Name attribute in the RADIUS authorization request will contain the remote IP address and if an account with such an account ID exists in the billing database and this account is allowed to call the dialed destination the call will be permitted to go through ip_auth table in porta sip database describes various ways to detect such SIP UAs It contains different patterns which may be applied to various parts of an incoming INVITE request if a certain pattern matches then IP authentication will be used PortaSIP may initiate IP authentication if any of the following match a pattern User Agent SIP header Remote IP address th
21. must support the 3 way calling feature Toll Restriction Feature description Blocks a station from placing calls to telephone numbers that would incur toll charges Provided using the tariff configuration in PortaBilling 700 900 Blocking Feature description Blocks a station from placing calls to 700 and 900 numbers Provided using the tariff configuration in PortaBilling 2000 2009 PortaOne Inc All rights Reserved www portaone com 73 Porta SIP 4 How to 2000 2009 PortaOne I nc All rights Reserved www portaone com System Concepts 74 Porta SIP How to Configure my Cisco gateway to accept incoming SIP calls and terminate them to a telephony network Configuration of the Cisco gateway for SIP is not much more difficult than H323 First of all make sure that the rest of your system is configured properly that the gateway can place the outgoing calls and is able to communicate with the billing using RADIUS Codecs First of all make sure you have set up a list of codecs which are supported by your SIP agents on your GW Your actual configuration might differ but here is a good example which should work in most cases voice class codec 1 codec preference 1 g723r63 codec preference 2 g729r8 codec preference 3 g729br8 codec preference 4 g723r53 codec preference 7 g726r16 codec preference 8 g726r24 codec preference 9 g726r32 codec preference 10 g7llalaw codec preference 11 g7llul
22. networks In addition to conventional IP telephony services PortaSIP provides a solution to the NAT traversal problem and enhances ITSP network management capabilities It can be used to provide residential business and wholesale traffic exchange services 2000 2009 PortaOne Inc All rights Reserved www portaone com 9 Porta SIP System Concepts PortaSIP functions D 1 Termination Termination oo partner A partner B 3 0 1 2 Bank Online payment processor Porta Billing Porta SIP Ms Administrator interface S Residential IP Admin Web S Pre paid cards 4 ANI DNIS Termination Callback to PSTN Customized VR Porta E UM phaser ez C Unified Messaging Phone amp Web Interface PortaSIP provides the following functionalities e SIP registration allowing SIP phones to use the service from any IP address static or dynamically assigned e Customizable greeting upon successful service activation e Authorization for all incoming calls e Customer numbering plans to ensure correct phone number translation e Facilitation of communication between SIP phones behind a NAT e Error announcements from the media server e Automatic disconnect of calls when the maximum credit time is reached e Automatic disconnect of calls when one of the parties goes offline due to a network outage e Various IP Centrex features call waiting call hold music on hold abbreviated di
23. number in the UK for their UK based sales representative In general each additional phone number is provisioned as an account in PortaBilling and then a corresponding SIP phone is registered to PortaSwitch using this account ID to receive incoming calls But some IP PBXs e g SPA 9000 can only register a single telephone number account with the SIP server In this case you may set up calls from additional phone numbers to be forwarded to the main account using the follow me feature For example an IP PBX registers to PortaSwitch with account 12061234567 however DIDs 18007778881 and 4412345678 must also be delivered to the IP PBX So you would set up accounts 18007778881 and 4412345678 with follow me to 12061234567 All calls will then be correctly routed to the IP PBX however since they all arrive to the IP PBX as calls to 12061234567 calls to different DIDs cannot be distinguished e g if a customer originally dialed the 1800 number he should be connected to general sales while if the UK number is dialed the call should be answered by a specific sales team group In this situation when defining a forwarding destination you should also activate the Keep Original CLD option available in advanced forwarding mode This will instruct PortaSwitch that the call must be forwarded to destination 12061234567 in this case to a registered SIP phone with this number while the To in the INVITE message should contain the original DID The I
24. of all you must record a set of all the required voice prompts account_expired cld_blocked and others Convert them into raw format and name the files lt original name gt lt language gt sln for instance the Chinese version of the account expired message will be contained in the file account_expired ch sin Upload the files to the PortaSIP server in the usr local share asterisk sounds directory This will be sufficient to enable the PortaSIP media server to play this voice prompt to SIP phones using g711 GSM and many other popular codecs Unfortunately you cannot perform such online transcoding into the g723 of g729 codec since in this case you must pay a license fee A solution is to pre convert this voice prompt into a g723 or g729 byte stream store it in a file with the same name but with the 723 or g729 extension and upload it to PortaSIP The media server will then use the appropriate file calculate how much bandwidth I need for my PortaSIP server The amount of bandwidth required for SIP signaling is insignificant compared to that used by the RTP stream so the most important task is to correctly estimate your RTP bandwidth needs of course this is only applicable if an RTP proxy is used otherwise the voice stream goes directly between the SIP phone and the remote gateway The 2000 2009 PortaOne I nc All rights Reserved www portaone com 81 Porta SIP How to http www voip info org wik
25. rules or abbreviated dialing table 121 is converted to 12027810009 o Checks if A is actually allowed to call that number and what is the maximum allowed call duration o Checks whether the dialed number is one of our SIP accounts if it is currently registered and what is the NAT status of both SIP phones Based on the results of the above operations billing sends an authorization response to the SIP server 3 e The SIP server checks its registration database to find the actual contact address of the SIP user agent with that number 2000 2009 PortaOne Inc All rights Reserved www portaone com 12 Porta SIP System Concepts e The SIP server sends an INVITE to the SIP user agent for user B 4 e If one of the SIP phones is behind NAT the SIP server will be instructed by the billing to send a voice stream via the RTP proxy Otherwise the SIP server may allow A and B s user agents to talk directly to each other e When the call is finished the SIP server sends accounting information to the billing The called party is not online Porta K Billing Porta UM Unified Messaging SIP phone A SIP phone B Offline or Not Answering e User A dials 121 in an attempt to reach user B His SIP user agent sends an INVITE request to the SIP server 1 e The SIP server performs authorization in the billing 2 The billing will perform number translation and determine whether the destination number is actual
26. the RTP proxy on a separate server SIP Signaling This server is installed next to the PortaSIP server running the SIP proxy and B2BUA Since fast and reliable connectivity between the PortaSIP server and the RTP proxy is important they should be connected by the same Ethernet link The RTP proxy will be controlled by B2BUA so that when a customer tries to make a phone call and proxying is required 2000 2009 PortaOne Inc All rights Reserved www portaone com 21 Porta SIP System Concepts B2BUA will give a command to open the ports and the RTP stream will flow to the server running the RTP proxy PortaSIP Presence Server PortaSIP enables IP Telephony Service Providers to deliver a presence service that allows users to monitor each other s availability and make decisions about communicating Presence information is highly dynamic and generally indicates whether a user is online or offline busy or idle away from or nearby a communication device and so on Having real time information about presence lets you increase the effectiveness of your communication and enjoy greater flexibility when setting up short term meetings and conference calls In other words it can save you time and money Today nearly all VoIP multimedia clients such as eyeBeam x Lite and MS Messenger support presence services In order to provide such services i e to handle presence requests a PortaSIP presence server is required This se
27. the session protocol sipv2 dial peer voice 101 voip description Incoming SIP calls incoming called number 2000 2009 PortaOne Inc All rights Reserved www portaone com 77 Porta SIP How to voice class codec 1 session protocol sipv2 dtmf relay rtp nte fax protocol cisco configure my Cisco ATA186 to work with PortaSIP Perform the initial network configuration of the ATA using the built in IVR After your ATA is assigned an IP address you can go to the web configuration screen at http lt your ATA I P address gt dev Consult Error Reference source not found For other options not listed in the table below the default manufacturer value is assumed provide services to and bill a customer who has a SIP enabled gateway but no authorization capability e g Cisco AS5350 PortaSIP is able to authenticate incoming calls using the IP address of the remote side This method ensures that PortaSIP will accept calls from yout own gateways but it can also be used to bill traffic from your customers You just need to create an account for your customer with an account ID identical to the IP address of his gateway Authentication and billing will be done in the same way as IP based billing using H323 make all SIP calls to a certain prefix NNN go to my gateway XXX Normally it is only possible to use the REGISTER command for user agents i e for devices which represent a single physical phone An SIP
28. ASTER above RADIUS secret key for RADIUS requests to the billing value string Port on the RADIUS server to which authentication requests should be sent 1812 by default Port on the RADIUS server to which accounting requests should be sent value number 1813 by default How long the SIP server should wait for a reply from the RADIUS server before retransmit value number 3 by default How many retransmit attempts should be made value number 5 by default Special features configuration Variable FIRSTLOGIN_CLI FIRSTLOGIN_ENABLE Activate the first login greeting feature B2B_KA A INTERVAL Ifa non zero value X is provided this Description possible values 0 or 1 Appear as CLI ANI number on the SIP phone for the first login greeting call value E 164 phone number enables sending keep alive requests to the caller party originating SIP device every X 2000 2009 PortaOne Inc All rights Reserved www portaone com 89 Porta SIP Administration FAQ B2B_KA O_INTERVAL SEND_START_ACCT MAX CREDIT_TIME HUNT_STOP REG_EXPIRES_MIN REG_EXPIRES_MAX ALLOW_ASYMMETRIC NO_VOICE_REJECTS YU_TEL_REMOVE PROCESS_REFER seconds zero value disables the keep alive packets If a non zero value X is provided this enables sending keep alive requests to the caller party terminating SIP device every X seconds zero value disables the keep alive packets Send an ac
29. EC http www faqs org rfcs rfc3489 html states This protocol is not a cure all for the problems associated with NA T STUN is merely a service that can be installed on a server such as PortaSIP allowing a STUN enabled SIP phone to communicate with it and detect the type of firewall it is behind and the public IP address of the NAT router Thus a SIP phone may obtain certain information by communicating with a STUN server but this will not have any effect on the way NAT handles IP packets traveling to or from the phone In the case of a cone firewall STUN information may help the SIP phone to determine in advance which IP address and port the remote party can use to communicate with it However in the case of a symmetric NAT this will not work and so an RTP proxy is still required Moreover since this 2000 2009 PortaOne Inc All rights Reserved www portaone com 87 Porta SIP Administration FAQ is a relatively new technology many phone vendors have not implemented the STUN functionality in its entirety or completely correctly So theoretically STUN may be used in conjunction with PortaSIP s RTP proxy if a phone detects that it can bypass NAT via STUN it will act as if it were on a public IP address and the RTP proxy will not be engaged Unfortunately in practice activating STUN only makes matters worse due to flaws in STUN implementation for IP phones Using two different approaches to handling NAT conc
30. However the RTP proxy method is the preferred solution due to the fact that it allows you to provide service regardless of the type or configuration of SIP phone and NAT router Thus you can say to customers Take this box and your IP phone will work anywhere in the world In general the smart method will only work if you are both an ISP and ITSP and so provide your customers with both DSL cable routers and SIP phones In this case they can only use the service while on your network 2000 2009 PortaOne Inc All rights Reserved www portaone com 36 Porta SIP System Concepts NAT Call Scenarios and Setup Guidelines With regard to NAT traversal there are several distinct SIP call scenarios each of which should be handled differently These scenarios differ in that in case 2 the media stream will always pass through one or more NATS as the endpoints cannot communicate with each other directly while in cases 1 and 3 it is possible to arrange things so that a media stream flows directly from one endpoint to another Calls between SIP phones 1 A call is made from one SIP UA SIP phone to another SIP UA SIP phone with both phones on public IP addresses outside a NAT In this case the phones can communicate directly and no RTP proxying is required A call is made from one SIP UA SIP phone to another SIP UA SIP phone and at least one of the phones is on a private network behind a NAT Here an RTP p
31. P Centrex features available in PortaSwitch as well as their activation and usage Please note that many of these features are either handled entirely on the IP phone or require adequate support from it such cases will be clearly indicated in the feature descriptions Also for your convenience we have provided instructions about how a particular feature can be used on an IP phone these instructions ate applicable to Sipura Linksys devices 1000 2000 2100 3000 For other types of IP phones please consult the manual provided by the vendor 2000 2009 PortaOne Inc All rights Reserved www portaone com 68 Porta SIP System Concepts Anonymous Call Rejection Feature description Automatically reject incoming calls from parties who do not deliver their name or telephone number with the call Provided by the IP phone dial the 77 code to activate this feature Automatic Line Direct Connect Hotline Feature description Automatically dials a pre assigned Centrex station s extension number or external telephone number whenever a user goes off hook or lifts the handset This feature is configured on the SIP phone side using the dial plan configuration parameter For example the following will implement a Hotline phone that automatically calls 1 212 5551234 SO lt 12125551234 gt The following creates a warmline to a local office operator 1000 after five seconds unless a 4 digit extension is dialed by the user 5
32. P PBX will then properly process incoming calls and will forward them to the correct recipient Selective Call Processing Sometimes incoming calls need to be treated differently calls from your boss or secretary should reach you on your cell phone even during the weekend while other calls can just go to voicemail Calls in the evening hours should go straight to your cell phone there is no point in ringing yout IP phone while you are not in the office while calls from your ex girlfriend should always go to voicemail All of this can be done using the selective call processing rules in PortaSwitch When the selective call processing feature is enabled for an account phone line you can define a set of rules that will be applied to 2000 2009 PortaOne Inc All rights Reserved www portaone com 53 Porta SIP System Concepts every incoming call Each rule may include some of the following limitations e From Calling number condition You can specify a list of phone numbers for a caller ANI or CLI which satisfy this condition e g you can list extensions for your boss and secretary your home phone your wife s cell phone number and so on When specifying a phone number you can enter either the full number of a pattern e g all numbers starting with 1800 Also when listing your colleague s phone number i e another phone in your IP Centrex environment you can enter its short extension number instead of the complete
33. PORTA ONE Administrator Guide Maintenance Release 19 www portaone com Porta SIP System Concepts Copyright Notice amp Disclaimers Copyright 2000 2009 PortaOne Inc All rights reserved PortaSIP Administrator Guide May 2009 Maintenance Release 19 V 1 19 2 Please address your comments and suggestions to Sales Department PortaOne Inc Suite 408 2963 Glen Drive Coquitlam BC V3B 2P7 Canada Changes may be made periodically to the information in this publication Such changes will be incorporated in new editions of the guide The software described in this document is furnished under a license agreement and may be used or copied only in accordance with the terms thereof It is against the law to copy the software on any other medium except as specifically provided in the license agreement The licensee may make one copy of the software for backup purposes No part of this publication may be reproduced stored in a retrieval system or transmitted in any form or by any means electronic mechanical photocopied recorded or otherwise without the prior written permission of PortaOne Inc The software license and limited warranty for the accompanying products are set forth in the information packet supplied with the product and are incorporated herein by this reference If you cannot locate the software license contact your PortaOne representative for a copy All product names mentioned in this manual are for identi
34. a 3 minute call to Russia the call will be automatically disconnected after 5 or 3 minutes respectively Follow me vs redirect number What is the difference between the follow me and associated number formerly called redirect number properties of an account While both seem to serve a similar purpose redirect numbers had several drawbacks e Different gateways applications had different kinds of support for this feature For instance the default Cisco debit card application did not support this feature at all e Using only a single phone number as a parameter did not permit flexible services For this reason a new flexible robust solution was required and so the call forwarding feature was implemented in PortaSwitch The redirect number feature is now obsolete and information in the redirect number field is no longer used by PortaSwitch PortaBilling still returns the 2000 2009 PortaOne Inc All rights Reserved www portaone com 52 Porta SIP System Concepts associated number value in the h323 redirect number RADIUS attribute for backward compatibility and so it can still be used by some external applications e g TCL scripts on a Cisco gateway Forwarding with the original DNIS CLD Very often a company operating an IP PBX would purchase multiple phone numbers all of which were to be routed to the company e g the main office phone number is in the New York area but the company also has an 1800 number and a
35. aSIP server performs authentication based on the IP address 5 6 Since this call is made from our trusted node gateway GW NY 01 the call is authorized e PortaSIP checks if the SIP user agent of the dialed number 12027810003 is registered at the time If yes a call setup request is sent 7 e If the dialed number belongs to an SIP account with unified messaging services enabled but this account is not online at the moment ot does not answer the call will be redirected to a voicemail system e After the call is completed the B2BUA sends accounting information for the two VoIP call legs to the billing The gateway will also send accounting information about the answet Telephony and originate VoIP call legs The billing engine will combine this information since accounting from the SIP server allows us to recognize that the call was terminated directly to the SIP user agent and not to a vendor while accounting from the gateway will contain information as to which account should be billed for this call PSTN gt SIP via VoIP DID Provider In the previous section we discussed traditional PSTN gt SIP service when a call is delivered to your gateway via E1 T1 lines and then forwarded to a SIP phone Unfortunately this service scheme assumes direct interconnection with the telco that owns DID numbers Establishing such direct interconnections with every telco from which you would like to get phone numbers can be problem
36. address This is ideal for SIP trunking and similar services This improvement results in an over 20 decrease in call processing time No special configuration is required on the PortaSIP side but you should specify your PortaSIP server s port 5061 on your gateway IP PBX outgoing SIP proxy with IP address VolP from Vendor Connection In the case of incoming calls from a vendor via IP there is one further issue since the call reaches your network via the Internet potentially anyone could be attempting to send you a call in such a fashion PortaSwitch must be able to correctly authorize calls coming from your vendors otherwise these calls will be dropped yet only calls from a real vendor should go through 4 1 5 gt l siis Porta K Billing SIP phone e Someone dials a phone number assigned to your customer 1 e The vendor receives this call from the PSTN network and sends the call to your PortaSIP server 2 e PortaSIP sends an authorization request to the billing 3 using either a remote IP address or a SIP username as the verification parameter for more details about these two methods of authentication see the IP authentication chapter e PortaBilling will check whether this authorization request is related to a VoIP from vendor connection 4 In there is no match it assumed to be a normal call from one of your customers and the call will then proceed according to the standard algorithm 2000
37. aling follow me etc e Fail over routing a list of routes arranged according to cost preference and customer routing plan is supplied by PortaBilling1 00 e Forwards calls to the unified messaging service PortaUM if a SIP phone is not available 2000 2009 PortaOne Inc All rights Reserved www portaone com 10 Porta SIP System Concepts PortaSIP Components 2 Provisioning Server er o RADIUS Client ae z we RADIUS nt gt a RADIUS rats RTP SIP Porta Proprietary PortaSIP components e SIP Proxy Server The SIP Proxy Server performs a number of functions such registering SIP telephones dealing with NAT issues etc e Back To Back User Agent B2BUA The B2BUA SIP based logical entity can recetve and process INVITE messages as a SIP User Agent Server UAS It also acts as a SIP User Agent Client UAC determining how the request should be answered and how to initiate outbound calls Unlike a SIP proxy server the B2BUA maintains the complete call state Integrating B2BUA with PortaSIP ensures that every call made between endpoints off net on net etc is authorized authenticated and billed The system is also able to provide Debit billing i e to disconnect a call if the account balance falls below zero Also B2BUA can automatically disconnect the other call leg if the SIP phone goes offline due to a network problem e RTP Proxy The RTP Proxy is an
38. ance and PortaSIP software packages PortaSIP installation and configuration are automated and integrated within the main installation process This allows you to install a completely functional PortaSIP server from scratch in less than 15 minutes For detailed installation instructions please refer to the PortaSIP Installation Guide What s New in Maintenance Release 19 This release includes several new features and improvements e DoS Denial of Service attack protection in SIP proxy e Support for BLF Busy Lamp Field functionality on IP phones 2000 2009 PortaOne Inc All rights Reserved www portaone com 7 Porta SIP System Concepts 1 System Concepts 2000 2009 PortaOne Inc All rights Reserved www portaone com 8 Porta SIP System Concepts PortaSIP s Role in Your VolP Network SIP phone SoftPhone im Te p 4 Billing Engine ATA186 Router ne x T Porta SIP SoftPhone PortaSIP is a call control software package enabling service providers to build scalable reliable VoIP networks Based on the Session Initiation Protocol SIP PortaSIP provides a full array of call routing capabilities to maximize performance for both small and large packet voice networks PortaSIP allows IP Telephony Service Providers to deliver communication services at unusually low initial and operating costs that cannot be matched by yesterday s circuit switched and narrowband service provider PSTN
39. apacity 100 Hide CLI Mode Vendor Account Translation Rule s 2A FA Outgoing Rule sh lt 2 CLI Translation Rule e 2000 2009 PortaOne Inc All rights Reserved www portaone com 66 Porta SIP System Concepts The basic Caller ID mechanism works much as it does in the case of email The caller information has a From header field including the address For example From John Smith lt sip 1234 sip example com gt tag 0099 8877 which means that user John Smith with phone number 1234 is trying to initiate an outgoing call using the sip example com server In Clear Caller Info mode default PortaSIP replaces the display name in the From field of the outgoing INVITE request John Smith in the example above with Anonymous while the phone number is removed So the From header field will look like this From Anonymous lt sip sip example com gt tag 0099 8877 Alternatively if Use Private Headers mode has been selected for the outgoing connection the From field is unchanged however the following data is added to the SIP packet From John Smith lt sip 1234 sip example com gt tag 0099 8877 Privacy id P Asserted Identity lt sip 1234 sip example com gt 2000 2009 PortaOne Inc All rights Reserved www portaone com 67 Porta SIP System Concepts IP Centrex Features This section provides a general overview of various I
40. at for instance an account which is blocked attempts to make a call e The customer tries to make a call SIP proxy receives the INVITE request and sends an authorization request to the billing e PortaBilling determines that this account is blocked An authorization reject is returned to the SIP server In addition to the h323 return code a special attribute is sent back to the SIP server This attribute contains a description of the type of error in this case user_ denied e The SIP server receives the authorization reject from the billing However instead of just dropping the call it redirects the call to the media server including the error message as a parameter e The media server establishes a connection with the SIP UA It locates a voice prompt file based on the error type and plays it to the user After this the call is disconnected The media server and prompt files are located on the SIP server So as to avoid dynamic codec conversion there are three files for each prompt pem 723 and 729 These files are located in usr local share asterisk sounds and you can change them according to your needs Here is a list of the currently supported error types account_expired the account is no longer active expired as per the expiration date or life time e cld_blocked there was an attempt to call a destination which is not in the tariff or is marked as forbidden e cld_dial_error a mistake was made when d
41. ate the customer s address This API may be different for different providers for instance Intrado uses an XML interface PortaBilling uses a plugin specific to each E911 vendor e Delivering a 911 call to the E911 provider network The actual method of interconnection depends on the provider e g via SIP of connection to a provider via PSTN trunks In PortaSwitch both these interconnection methods are configured using the standard routing tools 2000 2009 PortaOne Inc All rights Reserved www portaone com 44 Porta SIP System Concepts 2 Advanced Features 9 PortaOne Inc All rights Reserved www portaone com Porta SIP System Concepts IP Centrex Feature Management Convenient and efficient service provisioning is very important when you are managing an IP Centrex hosted IP PBX environment with tens or even hundreds of IP phones If you need to change a certain parameter e g CLI number for outgoing calls for all IP phones you will naturally want to avoid a situation in which you have to change this parameter manually for every account PortaSwitch divides call feature management into two parts e Some parameters are defined on the customer level and so are global for the customer s whole IP Centrex environment e Call features can also be managed on the account level You have the option of either manually overriding a certain parameter s value or specifying that the current value defined at the customer
42. atic e g if you want to give your customers the ability to choose a phone number from any European country you will need many gateways in different places Fortunately however there are more and more companies which offer incoming DID service i e they already have an interconnection with a specific telecom operator and so can forward incoming calls on these numbers to you via IP Thus no extra investment is required to provide phone numbers from a certain country or area except signing a contract with such a DID consolidator 2000 2009 PortaOne Inc All rights Reserved www portaone com 19 Porta SIP System Concepts X Telecom Vendor Phone C SIP phone A e C wishes to call A on his German phone number He thus dials A s phone number since C is in the US he dials it using the North American format 0114929876543 e The call is routed through the telecom network to the gateway of DID consolidator X Telecom 1 X Telecom in turn forwards this call to your PortaSIP server 2 e PortaSIP receives an incoming VoIP call and sends an authorization request to the billing 3 e The billing detects that this call is coming via a VoIP from Vendor connection so it initiates a special authorization for this call the call will be billed to the account which receives it Thus the maximum call time duration is calculated based on A s current balance e In the authorization response PortaSIP is instructed
43. aw codec preference 12 g723ar53 codec preference 13 g723ar63 SIP agent Now enable the SIP agent functionality on your gateway Also enable it on gateways where NAT symmetric traversal is supported as this will facilitate calls from SIP agents behind the firewall sip ua nat symmetric check media src NOTE Cisco GWs are currently unable to log in to the SIP server using the REGISTER method Dial peers Finally create an SIP enabled incoming dial peer dial peer voice 100 voip incoming called number T voice class codec 1 session protocol sipv2 dtmf relay rtp nte 2000 2009 PortaOne Inc All rights Reserved www portaone com 75 Porta SIP How to Note that this gateway provides no authentication of incoming SIP calls so that potentially anyone could route calls to you from their SIP server This is why the recommended configuration is as follows call application voice remote_ip flash app_ remote _authenticate tcl dial peer voice 100 voip incoming called number T voice class codec 1 session protocol sipv2 dtmf relay rtp nte application remote_ip Thus every incoming call will be authenticated by the IP address of the remote peer Since signaling for the SIP call comes from the SIP server this would be the address of the SIP server This means that calls coming from your own SIP server will be authenticated by billing since your SIP server is entered in the system as a trusted node configure m
44. be forwarded to voicemail To answer the ringing call appearance or originate a second simultaneous call the user simple puts the first call appearance on hold Calls on different appearances can be combined together to form a three way conference call Supported by PortaSwitch via the follow me feature The primary phone number account is provisioned on the IP phone and all the other appearances are created as accounts with the follow me configured to the primary account Music On Hold Feature description Provides a musical interlude for callers who are waiting on hold Supported by PortaSwitch every Centrex user can upload his own melody or use the default one for his Centrex environment Selective Call Acceptance Selective Call Acceptance SCA is a telecommunications system feature that allows customers to create a list of phone numbers from which they are willing to accept calls Supported by PortaSwitch via the Call Processing module every Centrex user can create rules defining a set of phone numbers If an incoming call matches one of these numbers the call is accepted otherwise the call is rejected 2000 2009 PortaOne Inc All rights Reserved www portaone com 72 Porta SIP System Concepts Selective Call Forwarding Selective Call Forwarding SCF is a telecommunications system feature that allows customers to forward callers from a selected group of numbers to another number Supported by PortaSwitch via the Ca
45. calls From a billing perspective a forwarded call is treated as two separate calls Thus if party A calls party B and B has follow me set up for phone number C the following will occur 1 PortaBilling will check if A is authorized to call B and for how long based on A s rates and the funds available in A s account 2 If forwarding is currently active on B s account PortaBilling will check if B is authorized to call C and for how long based on B s rates and available funds 3 After the call is completed the two accounts are charged and CDRs are produced accordingly one for account A for a call to destination B the other for account B for a call to destination C For A this call looks like any other call made to B If B is a number in the US it will look like a call to the US and A will be charged according to US rates even if the call was actually sent to a mobile phone in the Czech Republic For B the forwarded call is authorized and billed according to the same rules as a normal outgoing call from this account or you can apply a different rate plan for forwarded calls For instance if B is allowed to make outgoing calls only to US amp Canada and tries to set up a follow me number to India the number will not be usable If multiple follow me numbers have been defined each one will be authorized independently So if B currently has 1 available and this is enough to make a 5 minute call to the Czech Republic or
46. class can then be turned on off for an individual account Paging Intercom Calls Intercom calls enable users belonging to the same group to use two phones like an on door speakerphone When one user dials a special code before the other user s phone number a two way audio channel is established automatically The other user does not need to pick up his handset instead speaker phone mode is activated and both users can now talk to each other Most VoIP phones with the SIP protocol can be used for intercom calls Placing an intercom call Porta 4 Billing SIP phone A SIP phone B e User A dials an intercom prefix followed by User B s phone number His SIP user agent sends an INVITE request to the PortaSIP server 1 2000 2009 PortaOne Inc All rights Reserved www portaone com 61 Porta SIP System Concepts e An authorization request is sent to PortaBilling 2 e PortaBilling performs several operations o Checks that such an account exists and is allowed to use SIP services o Checks whether account B belongs to the intercom group under the same customer o Checks if the account is registered Based on the results of these operations PortaBilling sends an authorization response to the PortaSIP server with a special auto answer trigger 3 e The PortaSIP server adds the auto answer header to the outgoing INVITE request and sends the call to SIP user agent B 4 e The two call legs A and B
47. coming calls configure my Cisco gateway for PSTN gt SIP service Obtain a PSTN2SIP application Create an application and a dial peer to process incoming PSTN calls call application voice pstn2sip flash pstn2sip tcl call application voice pstn2sip authenticate by dnis call application voice pstn2sip skip password yes call application voice pstn2sip authorize yes call application voice pstn2sip dial account id yes dial peer voice 100 pots incoming called number T application pstn2sip voice port 0 d 1 The example above is for when you receive incoming calls with phone numbers already in E 164 If the number is received in a local format you will have to use the translate feature in the PSTN2SIP script to convert the number into E 164 For instance if you receive a US phone number in NANP area code phone number you should add the following command to the application configuration call application voice pstn2sip translate 1 Then configure your gateway to send outgoing calls to the SIP server according to the instructions in the previous topic support incoming H323 and SIP calls on the same gateway This configuration is supported as Cisco GW can handle both H323 and SIP calls at the same time However please note that Cisco matches an incoming dial peer by the incoming called number not by the protocol Thus the dial peer shown below will match both incoming SIP and H323 calls even if it gives
48. counting request to the billing when the call is started this is necessary if you want to display a list of active calls on the billing s web interface possible values 0 ot 1 Limit maximum call duration for all calls to a specified number of seconds value number 1 means unlimited List of SIP error codes which will stop hunting 1e trying the next route in the sequence value comma separated list of numbers Minimal interval between registrations in seconds defaults to 300 This parameter can be used to prevent hammering the SIP server with registrations every second Of so Maximum time interval during which the registration will be considered valid in seconds defaults to 7200 0 or 1 1 forces an RTP asymmetric flag for any non NAT UA The default is 0 0 or 1 if set to 1 prohibits forwarding of SIP UA to a media server for an error announcement if a problem is encountered e g incorrect password or invalid called number Useful if the PortaSIP instance is working in wholesale traffic exchange mode 0 or 1 forces B2BUA to remove options after in the userpart of CLD Such unusually formatted CLD may be sent by some types of network equipment 0 or 1 do internal processing of REFER requests After you have modified the porta sip conf file for a certain SIP instance you must restart that instance sudo var sipenv lt ip gt etc rc d sip sh restart 2000 2009 PortaOne Inc All rights Re
49. ctly reachable from a WAN to send information to and receive it from hosts on the WAN This is done with the help of the NAT server which is connected to the WAN by one interface with a public IP address and to the LAN by another interface with a private address This document describes issues connected with the implementation of NAT and its implications for the operation of PortaSIP with an overview of some fundamental NAT concepts The NAT server acts as a router for hosts on the LAN When an IP packet addressed to a host on the WAN comes from a host on the LAN the NAT server replaces the private IP address in the packet with the public IP address of its WAN interface and sends the packet on to its destination The NAT server also performs in memory mapping between the public WAN address the packet was sent to and the private LAN address it was received from so that when the reply comes it can carry out a reverse translation i e replace the public destination address of the packet with the private one and forward it to the destination on the LAN Since the NAT server can potentially map multiple private addresses into a single public one it is possible that a TCP or UDP packet originally sent from for example port A of the host on the private LAN will then after being processed in the translation be sent from a completely different port B of the NAT s WAN interface The following figure illustrates this here HOST 1 is a host
50. d SIP port e g 5060 SIP has tags which tell the proxy to do this The received tag tells the proxy to return a packet to a specific IP and the rport tag contains the port to return it to Note that SIP signaling should be able to traverse any type of NAT as long as the proxy returns SIP messages to the NAT from the same source port it received the initial message from The initial SIP message sent to the 2000 2009 PortaOne Inc All rights Reserved www portaone com 34 Porta SIP System Concepts proxy IP port initiates mapping on the NAT and the proxy returns packets to the NAT from that same IP port This is enabled in any NAT scenario Registering a client which is behind a NAT requires either a registrar that can save the IP port in its registration information based on the port and IP that it identifies as the source of the SIP message or a client that is aware of its external mapped address and port and can insert them into the contact information as the IP port for receiving SIP messages You should be careful to use a registration interval shorter than the keep alive time for NAT mapping RTP Media Stream An RTP that must traverse a NAT cannot be managed as easily as SIP signaling In the case of RTP the SIP message body contains the information that the endpoints need in order to communicate directly with each other This information is contained in the SDP message The endpoint clients fill in this info
51. d routers as well as a typical configuration for Cisco IOS software and Cisco ATA 186 telephones which has been adapted for optimal NAT traversal performance PortaOne RTP Proxy This provides an effective NAT traversal solution according to the RTP proxy method described above The RTP proxy is fully controlled by PortaSIP and is absolutely transparent to the SIP phone The RTP proxy does not perform any transcoding and so requires a minimum amount of system resources for call processing A RTP proxy in the dedicated mode on an average PC server can support about 500 simultaneous calls During the call initiation phase PortaSwitch gathers information about the NAT status of both parties caller and called participating in the call and decides about RTP proxying SIP to SIP calls S call 1 cai 1 7 7 Porta KASIP Si call 2 call 2 1 gt d I pe e 1 call 1 Phone 9 N 8 1 1 4 NAT 2 Phone B _ 7 NAT 1 For a SIP phone the possible conditions are e SIP phone on a public IP address e SIP phone behind NAT Thus the RTP proxy engagement logic for SIP 2 SIP calls can be summarized as follows e If both phones are on public IP addresses do not use an RTP proxy rather allow the media stream to go directly between them e If both phones are behind the same NAT router do not use an RTP proxy rather allow the media stream to go directly between them 2000 2009 PortaOn
52. e The following applies to PSTN gt SIP calls which you receive via a PSTN Oo gateway on your network For PSTN gt SIP calls received directly to your SIP server via VoIP see the next section In order to properly bill a SIP account for such calls do the following e Install a PSTN2SIP application on your Cisco gateway which handles incoming PSTN calls e Create an appropriate tariff with the desired rates For example if yout SIP customer has account 12021234567 and you want to charge him for incoming calls from PSTN to that number there should be a rate with a prefix matching this number for example 1202 e In the product associated with this account add an accessibility entry with this PSTN SIP gateway as the node and the tariff created in the previous step Now calls originating from a SIP phone to 1202 numbers will be charged using the tariff associated in the product s accessibility with the PortaSIP node Calls terminated from the PSTN to the SIP phone will be charged using a different tariff one associated with the PSTN gateway bill using different rate plans for incoming outgoing and forwarded calls This is done by assigning different access codes to entries in the product s accessibility e INCOMING This tariff will apply to calls to the PortaSIP server arriving from outside your network and terminated to one of your SIP phones e FOLLOWME This tariff will apply to forwarded calls e OUTGOING Th
53. e Inc All rights Reserved www portaone com 38 Porta SIP System Concepts e Otherwise the RTP proxy is used SIP to PSTN or PSTN to SIP calls If the called or calling party is a remote gateway or remote SIP proxy its NAT traversal capabilities are described in the PortaBilling configuration under connection properties The possible values are e Optimal This connection supports NAT traversal so it can communicate with an IP phone behind NAT directly This is the best possible scenario since you can entirely avoid using an RTP proxy when exchanging calls with this carrier e OnNat This connection does not support NAT traversal Direct communication with an IP phone is possible only if that phone is on a public IP address e Always Regardless of NAT traversal capabilities you must always use an RTP proxy when communicating with this carrier This may be necessary if you do not want to allow them to see your customer s real IP address or perhaps simply because this carrier has a good network connection to your SIP server but a poor connection to the rest of the world Thus you will need to proxy his traffic to ensure good call quality e Direct Always send a call directly to this gateway and never engage an RTP proxy PortaSIP cannot detect whether a remote gateway supports Comedia extensions symmetric NAT traversal If you do not use your own gateway for termination you should clarify this matter with your
54. e able to handle will be 60 provided that you have enough free DSPs in the system have problems with the audio quality of SIP calls what can I do First of all please make sure that both the user agents and SIP lt gt PSTN gateway are configured for use of the same low bitrate codec such as G 723 In APPENDIX B Cisco GW Setup for PortaSIP COMEDIA there are details on how to configure Cisco IOS and some models of IP phones for other SIP phones or gateways check the documentation supplied with the device If you are sure that the codec used for SIP calls is a low bitrate one for example by inspecting the gateway logs but the quality is still suboptimal you need to determine where packet loss is occurring in the media path To do this you can use standard network tools such as ping traceroute and the like Keep in mind that for SIP UA lt gt PSTN calls the RTP audio stream flows directly between SIP UA and PSTN GW while for SIP UA lt gt SIP UA calls the RTP path depends on whether or not an 2000 2009 PortaOne Inc All rights Reserved www portaone com 86 Porta SIP Administration FAQ RTP proxy is enabled If an RTP proxy is not enabled the RTP flows directly from one SIP UA to another Otherwise each RTP packet sent by one UA goes first to the machine running PortaSIP and is then resent from that machine to another SIP UA tried to register with the SIP server but my UA says registered even if my us
55. e address from which the INVITE request is received e Any of the SDP fields By default the following SIP UAs are considered incapable of digest authentication so that IP authentication is applied 2000 2009 PortaOne Inc All rights Reserved www portaone com 58 Porta SIP System Concepts e Cisco VoIP gateway any Cisco gateway running IOS this does not apply to Cisco ATA 186 188 Nextone SBC Sonus switch Mera SIP HIT Asterisk gateway Please ask the PortaOne support team for assistance in adjusting the information in this table to reflect the desired configuration of your network Multi DID Control If multiple DIDs sets of phone numbers have been allocated to a single user via the Account Alias feature the PortaSwitch administrator can define whether an alias is allowed independent SIP registration If the ability for authentication registration is turned off the alias cannot be provisioned on the IP phone or used for any other types of service activities Such an alias is used solely for the purpose of routing incoming calls to that DID to the main account This extends the available service options to hosted IP PBX and SIP trunking services If alias registration is allowed the alias can basically be used as another account Of course it still shares a balance with the main account This is useful for multiline telephones like SPA 941 where each line can have its own DID and be registered to PortaSIP indep
56. e world Auto provisioning This approach is a fundamentally different one Instead of attempting to contact an IP phone and change its parameters pop method the initiative is transferred to the IP phone itself The device will periodically go to the provisioning server and fetch its configuration file IP Phone Provisioning When you use auto provisioning for an IP phone instead of entering the same values for codec server address and so on into each of a thousand user agents you can simply create a profile which describes all these parameters Then PortaBilling can automatically create a configuration file for the SIP phone and place it on the provisioning server The only configuration setting which is required on the IP phone side is the address of the provisioning server i e where it should send a request for its configuration file When the IP phone connects to the Internet it will retrieve a specific configuration file for its MAC address from the TFTP or HTTP server and adjust its internal configuration If you decide later to change the address of the SIP server you need only update it once in the profile and new configuration files will be built for all user agents Each user agent will then retrieve this file the next time it goes online 2000 2009 PortaOne Inc All rights Reserved www portaone com 41 Porta SIP System Concepts Porta Billing Provisioning server
57. endently Call Parking Call parking allows users to put a conversation on hold and then resume it from a different IP phone Parking a call Porta M Billing SIP phone A SIP phone B 2000 2009 PortaOne Inc All rights Reserved www portaone com 59 Porta SIP System Concepts e A dials B s phone number 1 e An authorization request is sent to PortaBilling 2 if authorized successfully 3 the call is connected to B 4 e B requests that this call be parked by dialing a special call parking code 5 e The dialed code is sent to billing for verification 6 Upon successful approval 7 A is put on hold and hears the music on hold melody uploaded by B 8 e The call parking confirmation message is played to B 9 this message also contains information about the code to retrieve the parked call Retrieving a parked call Porta M Billing SIP phone A SIP phone B e Ais still connected via call parking 0 e B dials the retrieval code from any IP phone 1 e An authorization request is sent to PortaBilling 2 which determines that this is an attempt to retrieve the parked call 3 e The two call legs A and B are joined together Call Barring Call barring allows you to prohibit outgoing calls to specific destinations The main difference between call barring and blocking destinations in a tariff is that the latter applies to all customers using a given tariff plan while call barring can
58. ername or password are incorrect is there a security breach in PortaSIP Of course PortaSIP does not really allow unauthorized clients onto your network If the SIP UA tries to register using an incorrect username or password or with an account which is blocked registration will not succeed However UA will still receive registration confirmation and this is why you see registered in the UA But if you try to make an outgoing call it will be diverted to the media server where the appropriate message will be played e g This account does not exist or Account is blocked This allows SIP registration s troubleshooting to be greatly simplified Keep alive functionality does not work with my XXX brand SIP phone Your SIP phone must correctly respond to keep alive re INVITE requests If it does not support this functionality then it may either not reply at all to these requests or even worse assume that this is a new incoming call If PortaSIP detects that the SIP UA has not answered the first keep alive at the very beginning of the call when the SIP phone should presumably be online then it assumes that the SIP UA does not support this functionality and disables keep alives for this session In any case it is recommended to choose a SIP UA which supports re INVITEs e g Sipura I do not want to use an RTP proxy since it will increase the amount of required bandwidth can I use STUN instead The STUN R
59. erson you want to contact Type the person s complete e mail address fi 604 521 5277 Select the service that this person uses siP Communications Service s APPENDIX E SJPhone Configuration for PortaSIP 1 First you need to have the SJPhone installed on your machine After the installation start the SJPhone software and the following login screen will be displayed G Service PortaOne Please enter this information to initialize the service profile Lax Account 123456789 Password LLITTTTTT Save service information permanently 2000 2009 PortaOne Inc All rights Reserved www portaone com 99 Porta SIP Appendices 2 Key in the Account ID and password for the PortaSIP and press OK SJ Phone display should be similar to the one in the following snapshot showing the account balance in Ready to call state The phone is ready to be used 3 Right click on the softphone and press Login to change or make corrections to the Account Password 2000 2009 PortaOne Inc All rights Reserved www portaone com 100 Porta SIP APPENDIX F SIP Devices with Auto provisioning Currently PortaSwitch can auto provision the following SIP phones ATAs e Cisco ATA 186 firmware versions 2 and 3 e Sipura 1001 e Sipura 2000 e Sipura 2100 e Sipura 3000 e Linksys PAP2 e Linksys RTP 300 e Linksys Sipura SPA 2102 e L
60. es or adaptors which are distributed to your customers The MAC address parameter is essential for every IP phone which is to be automatically provisioned and so a corresponding entry must be created in the IP phone inventory 2000 2009 PortaOne Inc All rights Reserved www portaone com 42 Porta SIP System Concepts PortaSIP and E911 Services One of the most popular types of VoIP services provided by PortaSwitch is the residential telephony service including a substitute for a traditional PSTN line using a VoIP adaptor Here the issue of emergency services becomes very important since customers may not fully switch to a VoIP service provider unless it is resolved In most countries ITSPs are required to provide emergency services to their customers by the local authorities e g the FCC in the US Using PortaSwitch an ITSP can meet all such requirements and start providing residential or business IP telephony services PortaSwitch offers an FCC compliant framework for providing E911 services There are several components of E911 services e Subscriber and subscriber address The subscriber is the person who is using the telephony service and his address is his physical location to which the police fire department ambulance should be sent in case of emergency e An ITSP is a company providing telephony services to the subscriber e PSAP Public Safety Answering Point is an agency responsible for answering emergency calls in
61. ession in the SPA by typing http lt spa ip address gt admin advanced 3 Choose the specific phone port click on Line 1 Line 2 or another tab 4 Provide values for the required parameters which include a in Proxy and Registration i Proxy PortaSIP address or hostname ii Register yes b in the Subscriber information part i Display Name your identification e g John Doe this will be seen by the called party ii User ID SIP account ID ii Password VoIP password for your SIP account iv Use Auth ID no 5 Submit all the changes and update the SPA configuration 2000 2009 PortaOne Inc All rights Reserved www portaone com 94 Porta SIP Appendices 2000 2009 PortaOne Inc All rights Reserved www portaone com 95 Porta SIP Appendices Network Settings SIP TOS DiffServ Value RTP TOS DiffServ Value SIP Settings SIP Port EXT SIP Port SIP Debug Option Call Feature Settings Blind ttn Xfer Enable xfer When Hangup Conf Proxy and Registration Proxy Outbound Proxy Register Register Expires Use DNS SRY Proxy Fallback Intvl Subscriber Information Display Name Password Auth ID Mini Certificate ISRTP Private Key Call Waiting Serv Block ANC Serv Cfwd All Serv Cfwd No Ans Serv Cfwd Last Serv Accept Last Serv ICID Serv Call Return Serv Three Way Call Serv Attn Transfer Serv 0x68 Oxb8 5060
62. fication purposes only and are either trademarks or registered trademarks of their respective owners 2000 2009 PortaOne Inc All rights Reserved www portaone com 2 Porta SIP System Concepts Table of Contents Preface arr A EI E A E E E A RT 5 Hardware and Software Requirements 6 11588113110101 7 What s New in Maintenance Release 197 7 1 System 1 1 1 1 1 1 1 1 1 1 1 1 1 1 2 2 2 2 2 2 2 8 PortaSIP s Role in Your VolP Network 9 Portasi P COoMpPOneENtSs 11 Call Process Supported Services 12 Separate RTP Proxy Server 21 PortaSIP Presence Server 22 Instant Messaging 23 Virtual SIP Servers a E E 25 Clustering of PortaSIP Servers 26 Call Flow Scenarios for a PortaSIP Cluster 28 Understanding SIP Call Routing 32 NAT Traversal Guidelines 33 Auto provisioning IP Phones 40 PortaSIP and E911 Services 43 2 Advanced F AUTSS ir cames 45 IP Centrex Feature Management 46 6311113105121 nn nn lanta 46 Call 49 Selective Call Processing 53 Service Announcements via the Media Server 55 NAT 1 6 6 anses aa 56 Keep alive Call Monitoring 57 First Login Greeting
63. he service via other SIP servers Better Network Utilization You can install several SIP servers in different geographical locations as shown below enabling users within a certain network to use the closest available SIP server So if user A from Singapore calls user B also from Singapore the call will be handled by the PortaSIP server in Singapore and the voice traffic will travel only via the Singapore backbone xO 57 Re kene Pore ITSP a PortaSIP 5 S V a 4 PortaSIP wR PortaBilling PortaSIP a 5 A A Master Slave 2000 2009 PortaOne Inc All rights Reserved www portaone com 27 Porta SIP System Concepts This allows VoIP services to be efficiently provided in a situation which is highly typical for many countries or regions good fast Internet connectivity inside the country region and mediocre connectivity with the rest of the world For all users inside that region VoIP traffic signaling and RTP will travel on the local backbone while only small RADIUS packets will travel to the central PortaSwitch location Call Flow Scenarios for a PortaSIP Cluster SIP UA lt gt SIP UA Case A Both SIP phones are registered to the same PortaSIP server SIP PortaSIP PortaSIP Porta K Billing 7 Billing Provisioning XS 1 1 1 Billing Engine In this case the call flow is exactly the same as in a situation where only one PortaSIP server is available disc
64. hen proceeds in basically the same way as if it were communicating directly with C s SIP user agent e After the call is established the B2BUA starts the call timer disconnecting the call once the maximum call duration is exceeded e After the call is completed the B2BUA sends accounting information for the call to the billing 2000 2009 PortaOne Inc All rights Reserved www portaone com 16 Porta SIP System Concepts Terminating SIP calls to a vendor using telephony Porta M Billing SIP phone A GW NY 02 Phone C 12 34 56 78 Let s assume that Tl is connected to Qwest on our gateway GW NY 02 in New York where we are able to terminate calls to the US This connection would be described as a PSTN to vendor connection The PortaSIP server obtains the address of the GW NY 02 gateway in the route information e The B2BUA sends an INVITE to the remote gateway GW NY 02 e GW NY 02 performs authentication on the incoming call via the remote IP address Even if the call was actually originated by A a dynamic IP address but the INVITE request to GW NY 02 atrived from the PortaSIP server the PortaSIP s IP address will be authenticated Since PortaSIP is defined as our node authentication will be successful NOTE Remote IP authentication on the gateway is not required in this case but is highly recommended Otherwise someone else might try to send calls directly to the gateway bypassing authentication and
65. his is the second PortaSIP server Note that although SIP signaling will travel via both SIP servers this is not the case with RTP voice traffic Depending on the NAT context of the call and the RTP proxy configuration PortaSwitch may either connect the RTP stream between the phones directly or use the RTP proxy on one of the SIP servers So even if two SIP servers are involved in this call this does not affect call quality since the RTP stream follows the standard path SIP phonel gt SIP server gt SIP phone2 2000 2009 PortaOne Inc All rights Reserved www portaone com 29 Porta SIP System Concepts SIP UA gt PSTN When a SIP phone user makes a call to an off net destination only one PortaSIP server and PortaBilling are involved in the call flow So this works in exactly the same way as described earlier for SIP gt PSTN calls in the case of a single PortaSIP server Porta K Billing Billing Engine Billing Provisioning 1 I lt A PortaSIP 6 ee PSTN gt SIP UA Again the call flow is extremely similar to the usual PSTN gt SIP call flow The gateway delivers a call to a PortaSIP server which then sends the call to the SIP phone Porta K Billing Billing SIR Provisioning PortaSIP 2000 2009 PortaOne Inc All rights Reserved www portaone com 30 Porta SIP System Concepts SIP Phone Configuration for PortaSIP Cluster In order to ensure reliable VoIP se
66. hrough or no more routes are left e If such a call was completed to follow me number R two CDRs will appear in the system one for the call C gt A charged per the incoming rates for A and the other for C gt R charged per the outgoing rates for A e Ifthe call did not originate in the PSTN network but rather from user B s SIP UA two CDRs will likewise be generated B will be charged for call B gt A while A will be charged for call B gt R The follow me service can be recursive Thus A can forward calls from his SIP phone to B s SIP phone and B can forward calls to his mobile phone number C Note that in the case of such a multi hop follow me A gt B gt C gt D gt PSTN number only two CDRs will be produced similar to a simple follow me e a CDR for the caller billed to A A gt B e a CDR for the forwarder outside the network i e the last SIP account in the follow me chain billed to D A gt PSTN Simultaneous ringing You can define a follow me list with several phone numbers all of which will ring concurrently The first one to answer will be connected to the incoming call You can also include you own phone number on the list of phone numbers for simultaneous ringing Your IP phone will then ring together with the other phones e g your home phone or cell phone and you can answer either one of them In this case you are advised to modify the call processing so that it does not include the Ring action but
67. i Bandwidth consumption website provides information regarding bandwidth consumption by voice calls depending on the codec used Do not use the codec bitrate in your calculations but rather an actual bandwidth figure which takes IP headers into account For example if you anticipate a maximum of 60 simultaneous calls with the g729 codec you will need 31 2Kpbs 2 60 3 7Mbps Note that we multiply the one call bandwidth not just by the total number of calls but also by 2 since every call will be coming both in and out of the RTP pt OXY enable my SIP phone or ATA to be automatically provisioned by PortaSwitch First of all you must make sure that your device supports auto provisioning see APPENDIX F SIP Devices with Auto provisioning Then create the required IP phone profile and enter information about the IP phone into the inventory Provision the SIP service as described in this manual and then assign it to an available port on your IP phone in the account info screen for a SIP account Enter information about the provisioning server into your IP phone s configuration In some cases you may need to restart the IP phone in order to force a configuration update from the provisioning server 2000 2009 PortaOne Inc All rights Reserved www portaone com 82 Porta SIP Administration FAQ 5 Administration FAQ 2000 2009 PortaOne Inc All rights Reserved www portaone com 83 Porta SIP
68. ialing e cld_tmp_unavail the account you are trying to contact has configured the incoming call to be dropped or is out of money 2000 2009 PortaOne Inc All rights Reserved www portaone com 55 Porta SIP System Concepts e cld_unassigned the dialed number is configured to be terminated inside the network but has not been assigned to any particular user yet e credit_disconnect a call is disconnected because the maximum credit time is over in_use this call attempt is blocked because another call from the same debit account is in progress e insufficient_balance there are not enough funds to make a call to the given destination e invalid_account incorrect account ID or account is not permitted to use SIP services e user_denied the account is blocked e wrong passwd an incorrect password has been provided Every account in PortaBilling has a preferred language property which defines the desired language for IVRs The language code e g ch for Chinese assigned to the account is returned from the billing so the media server will first attempt to play a voice prompt for that language If that prompt does not exist the default English voice prompt will be played NAT Keep alive When a SIP phone behind NAT registers to the SIP proxy the NAT router creates an internal tunnel between LAN and WAN passing all communication for this network connection back and forth between the client
69. ication and replay protection to voice traffic between IP phones Voice VPN Rating The Voice VPN Virtual Private Network feature provides special handling of calls within a specific IP Centrex environment typically the telephony system for a certain enterprise Most of its features e g abbreviated dialing have been previously discussed but there is one important issue remaining how these calls will be charged We need to have a consistent way of charging all calls between a customer s IP phones regardless of the actual phone number dialed for instance the customer may have phone numbers from different countries When the Voice VPN feature is enabled for a particular customer and a call is made from account A belonging to this customer to account B belonging to this same customer PortaBilling will look up the applicable rate not for the actual phone number but for the special keyword VOICEVPN and use this to charge the call When entering a rate to that destination in the tariff applied to your customers you can specify how such calls are to be rated should they be free calls or charged a nominal amount and so on Using the vorcEvP rate in tariffs allows you to avoid having SIP to SIP minutes mixed in with off net minutes when products with volume discounts ate used One associated feature is Voice VPN Distinctive Ring When activated for a call arriving from any IP phone within the same IP Centrex environ
70. inksys SPA 941 e Linksys SPA 962 e Linksys WRT54GP2 e GrandStream GXW400x e GrandStream HT286 e GrandStream HT486 e GrandStream HT488 e GrandStream HT496 Appendices We are constantly working to extend the list of supported IP devices If the IP phone you plan to use is not listed here please contact us it may already be scheduled for a future release or we may include it at your request 2000 2009 PortaOne I nc All rights Reserved www portaone com 101
71. ioned on his phone plus some extra accounts e g 4981234567 with the follow me service on these accounts configured to always go to 12027810003 create an application to handle PSTN gt SIP calls You can create this application yourself according to the functionality description in this guide A PSTN2SIP application may be purchased by contacting the PortaOne sales team configure SIP phone X made by vendor Y Obviously we cannot provide a sample configuration for every possible SIP phone model Please check the documentation shipped with your device Essentially however you need to configure the following settings e IP address of the SIP proxy IP address or hostname of the PortaSIP server e CID Caller Identification e Login and password account ID and password of the corresponding account in PortaBilling 2000 2009 PortaOne Inc All rights Reserved www portaone com 79 Porta SIP How to e Preferred audio codec depends on your network characteristics should be compatible with the codec used by other components e g VoIP gateways used for PST N termination In the case of PortaSIP both the login name and CID should be set to the same value Set the preferred audio codec to G 723 if your phone supports this Likewise enable in band alerting if your phone supports it as this will help in situations when the phone is behind a NAT bill incoming calls from PSTN to SIP using a special rat
72. is tariff will apply to calls originating from IP phones Although you may specify OUTGOING as an access code it is recommended that you keep this entry as a default 2000 2009 PortaOne Inc All rights Reserved www portaone com 80 Porta SIP How to i e with an empty access code Then if further possibilities for different rate plans e g special rating for calls on hold are added in future releases this rate plan will be automatically applied to these new entries o 2 Edit SuperCall Product gt add Save Save amp Close Close Rate Lookup 422 Objects Logout B Log Product Name Supercall 5 Currency USD Managed By Administrator only General Info Maintenance Fee Online Signup Accessibility Subscriptions Notepad Edit Service Type Node Access Code Info Digits Tarif Delete Voice calls PortaSIP FOLLOWME SuperCall forwarded calls Voice calls PortaSIP INCOMING SuperCall incoming calls x Voice calls PortaSIP SuperCall outgoing calls The information above assumes that PSTN gt SIP calls arrive directly to your PortaSIP server If they arrive via the gateway on your network replace INCOMING with a row containing your PSTN gateway as explained in the previous topic provide error messages from the media server in my users local language First
73. isk space this space is required for storing various log files Intel Xeon or AMD Opteron processor running at 1 8 GHz or greater Additional processor speed is needed for networks with a high call volume Atleast 1 GB of RAM 2 GB recommended At least one USB port For information about whether particular hardware is supported by FreeBSD from the JumpStart Installation CD consult the related document on the FreeBSD website http www freebsd org doc en_US 1SO8859 1 books faq hardware html Client System Recommendations OS Windows 95 XP UNIX or Mac OS Browser Internet Explorer 6 0 FireFox 2 0 with JavaScript enabled Spreadsheet processor MS Excel Display settings o Minimum screen resolution 1024 x 768 o Color palette 16 bit color minimum NOTE To view downloaded CSV Comma Separated Values files in Windows please do the following to match PortaBilling s default list separator My Computer gt Control Panel gt Regional Settings gt Number gt List Separator type 2000 2009 PortaOne Inc All rights Reserved www portaone com 6 Porta SIP System Concepts Installation In order to simplify installation and support as much as possible PortaSIP is provided on a jump start installation CD This CD contains installation media for FreeBSD 6 3 stable branch with the latest security bug fixes supplementary packages necessary for convenient system administration and mainten
74. l Waiting Feature description Allows a caller s name and number to be displayed when the called party is taking another call Supported by PortaSwitch the phone must have a display to show the caller ID and the Call Waiting feature must be activated Consultation Hold Feature description Calls can be put on hold by depressing the switch hook or pressing the flash button After completing the second call the user is automatically reconnected to the original call on hold Supported by PortaSwitch Distinctive Ringing Feature description Uses a special ringing pattern to indicate whether an incoming call is from inside or outside the Centrex group Supported by PortaSwitch for the VPN Distinctive Dialing feature Group Pickup Feature description Allows phones in the same IP Centrex environment all accounts under the same customer to answer each other s calls by dialing a Group Pickup Pretix on their phones Supported by PortaSwitch Intercom Dialing Feature description Allows a receiving phone to auto answer a call and activate speakerphone mode Supported by PortaSwitch the Paging Intercom feature must be activated Hunt Groups Feature description Allows calls to be redirected to other predetermined lines when the line called is busy Hunting allows a number of lines to be grouped into a pool so that incoming calls are directed to whichever of these lines is available Supported by PortaSwitch via the fol
75. le This is not strictly required however and therefore some of them will just use a random source port for outgoing connections e Whether or not another session has already been established through the NAT from a different host on the LAN with the same source port In this case the NAT server is likely to allocate a random port for sending out packets to the WAN Please note that the term already established is somewhat vague in this context The NAT server has no way to tell when a UDP session is finished so generally it uses an inactivity timer removing the mapping when that timer expires Again the actual length of the timeout period is implementation specific and may vary from vendor to vendor or even from one version by the same vendor to another NAT and SIP There ate two parts to a SIP based phone call The first is the signaling that is the protocol messages that set up the phone call and the second is the actual media stream i e the RTP packets that travel directly between the end devices for example between client and gateway SIP signaling SIP signaling can traverse NAT in a fairly straightforward way since there is usually one proxy The first hop from NAT receives the SIP messages from the client via the NAT and then returns messages to the same location The proxy needs to return SIP packets to the same port it received them from i e to the IP port that the packets were sent from not to any standar
76. level should be used This allows you to define most call feature parameters only once on the customer level These will then be automatically propagated to accounts individual phones Call Transfer In a typical call transfer party A sends a SIP REFER message to party B and this causes party B to initiate a new call according to the parameters specified in the REFER message destination and the like While this works just fine with IP phones on your VoIP network it may not work in the case of SIP gt PSTN or PSTN gt SIP calls since you will not always know if your PSTN carrier supports REFER messages in fact many do not support it To eliminate this problem and allow your users to make call transfers anytime and anywhere PortaSIP will intercept the REFER message and process it entirely on the PortaSwitch side Every REFER message is authorized in PortaBilling So if A transfers a call to a phone number in India the billing will validate whether A is actually allowed to make this call and limit the call duration according to A s available funds After that PortaSIP will proceed to establish a new outgoing call and connect the transferred party When the call is finished A the party who initiated the transfer will be charged for the transferred portion of the call this applies regardless of whether A was the called or calling party in the original call This allows you to transparently charge call transfers and avoid fraudulent
77. ll Processing module every Centrex user can create rules defining a set of phone numbers If an incoming call matches one of these numbers the call is forwarded to the destination defined in the call forwarding or follow me settings Selective Call Rejection Selective Call Rejection SCR is a telecommunications system feature that allows customers to reject incoming calls Supported by PortaSwitch via the Call Processing module every Centrex user can create rules defining a set of phone numbers If an incoming call matches one of these numbers the call is rejected Speed Dialing Feature description Allows the user to dial frequently called telephone numbers using an abbreviated speed calling code instead of the entire number Supported by PortaSwitch via the Abbreviated Dialing feature Station Message Detail Recording SMDR Feature description Allows the corporate telecom manager to receive call detail records on a per station basis before the monthly telephone bill is even issued SMDR helps the customer control telephone fraud and abuse perform accurate cost accounting and analyze call patterns to identify opportunities for cost reductions Supported by PortaSwitch call details are available on the PortaBilling web interface Three Way Conferencing Three way calling Feature description Allows user to add a third party to an existing conversation forming a three way conference call Supported by PortaSwitch SIP phone
78. llow me services The follow me feature allows you to receive calls even if your IP phone is offline at the moment You can specify several alternative destinations for a single destination number account Follow me is activated when IP phone is offline not registered IP phone replies with an error code i e the line is currently busy because you are making another call No answer is received within a certain interval usually 20 seconds the phone may be online but nobody answers or there is a network outage For instance if you do not pick up your IP phone or the IP phone is unreachable due to a network error the call would be forwarded to your home phone if not answered within 30 seconds it would be forwarded to 2000 2009 PortaOne Inc All rights Reserved www portaone com 49 Porta SIP System Concepts your mobile phone and so on For each of these phone numbers you can define the period when a given phone should be used for example calls should be forwarded to your home phone only from 8 in the morning until 9 in the evening 2 Porta K4 Billing GW NY 01 Phone C SIP da A fn Le X SIP phone R e C wishes to call A So he dials A s phone number since C is in the US he dials it using the North American format 2027810003 e The call is routed through the telecom network to gateway GW NY 01 When the incoming call arrives at the gateway 1 it is processed there in exactly the same way as a no
79. low me feature 2000 2009 PortaOne Inc All rights Reserved www portaone com 71 Porta SIP System Concepts Message Waiting Audible Feature description Provides the user with an audible notification a stutter dial tone when messages have been left in the extension s voice mail system Supported by PortaSwitch the actual message waiting SIP info packet is originated by PortaUM and relayed by PortaSIP Message Waiting Visual Feature description provides the user with a visual indication when messages have been left in the company s voice mail system Supported by PortaSwitch the actual message waiting SIP info packet is originated by PortaUM and relayed by PortaSIP requires the phone to be able to display the appropriate icon Multiple Call Appearances Feature description Multiple Call Appearances allow each station to have two or more appearances of the user s primary phone number Each appearance gives the user the ability to handle one call Consequently Multiple Call Appearances allow the user to originate and or terminate multiple calls simultaneously Unlike an analog multi line phone the station needs only one line and one phone number for Multiple Call Appearances When the user is involved in a call on one call appearance and another call is offered on a different call appearance the user may use the Caller ID information to decide whether to answer the ringing call appearance or let the call
80. ly an account e The billing checks the registration database but finds that this account is not online at the moment If B has unified messaging services enabled the billing will return routing 3 for this call which will be sent to the UM gateway Thus A will be redirected to a voicemail system and can leave a message for B 6 The same thing would happen if B were online but not answering his phone 4 5 e In any other case the call will fail Call between several PortaSIP servers You can use several PortaSIP servers simultaneously for improved reliability or better network utilization Let s assume you have two PortaSIP servers the primary one in New York and a second one installed in Frankfurt The Frankfurt PortaSIP serves most of your European customers i e they connect to it via the fast intra European IP backbone and acts as a backup for all other users around the world Thus the SIP phone will try to register there if the New York server is down or for some reason inaccessible 2000 2009 PortaOne Inc All rights Reserved www portaone com 13 Porta SIP System Concepts Porta 4 Billing SIP phone A SIP phone B In the example above user A assigned SIP phone number 12027810003 and registered to PortaSIP in New York calls user B with phone number 4981234567 who is currently registered to PortaSIP in Frankfurt e A dials B s number 4981234567 His SIP user agent sends an INVITE request
81. making such calls for free e The call will be routed to the PSTN on the gateway e After the call is established the B2BUA starts the call timer disconnecting the call once the maximum call duration is exceeded e After the call is completed the B2BUA sends accounting information for the two VoIP call legs to the billing The gateway will also send accounting information about the answer VoIP and originate Telephony call legs The billing engine will combine this information since accounting from the SIP server allows us to identify who made the call while accounting from the gateway carries other useful information for example through which telephony port the call was terminated 2000 2009 PortaOne Inc All rights Reserved www portaone com 17 Porta SIP System Concepts PSTN gt SIP Porta Porta Billing Porta Billing Phone C g 2 SIP de A This is another important aspect of SIP telephony Your subscribers not only want to make outgoing calls they also want other people to be able to call them on their SIP regardless of where they are at the moment In order to do so you will need to obtain a range of phone numbers from your telecom operator and make sure that calls made to these numbers on the PSTN network are routed to your gateway via the telephony interface e C wishes to call A He thus dials A s phone number since C is in the US he dials it using the North American format 2027810003
82. ment PortaSIP will instruct the IP phone to use a ring pattern different from the default one the phone must support distinctive ringing This allows the end user to immediately recognize whether the call is coming from one of his co workers or from an external number Voice On net Rating By using VoIP technology and PortaSwitch Internet telephony service providers can truly make the world flat for their customers It is possible to reach phone numbers in virtually any country in the world and as easy to make a call to the opposite hemisphere as to your neighbor ITSPs wishing to offer special pricing for calls made between IP phones connected to PortaSwitch regardless of the actual phone number can use 2000 2009 PortaOne Inc All rights Reserved www portaone com 65 Porta SIP System Concepts the Voice On Net feature When enabled all calls between IP phones will be rated according to the special destination VOICEONNET So if customer A has a US phone number assigned to him and calls a phone number in India assigned to another customer in your system customer A will not be charged the international rate for this call but rather a special On Net rate defined by you Support for Caller Identity and Privacy Flags A user may sometimes indicate that he wants privacy for a particular outgoing call i e the other party should not see his phone number So when sending the call to a third party carrier PortaSIP must show the
83. n the server The messaging module is implemented as an internal part of the PortaSIP proxy server and enables communication between users by means of SIP MESSAGE packets Porta Ki Mu set IM Client 1 IM Client 2 A basic instant messaging flow will look like this e Users connect to PortaSIP with user agents IM clients e Users are identified by an address i e John Smith lt sip 1234 sip example com gt that uniquely defines an individual within PortaSIP e To make themselves available for contact via a particular SIP user agent users send a SIP REGISTER message to the PortaSIP proxy e Once users have been registered they can send MESSAGE requests to each other via the PortaSIP proxy e When a message reaches its destination a 200 OK response is returned This does not necessarily mean the message has been read by its recipient 2000 2009 PortaOne Inc All rights Reserved www portaone com 24 Porta SIP System Concepts Virtual SIP Servers On a single PortaSIP installation one physical server one license you can run multiple virtual PortaSIP instances each of them a separate server that can be used in a PortaBilling virtual environment The only thing required to create a new SIP instance on the SIP server side is adding an extra IP address IP alias and populating the configuration files Porta K Billing Porta K4 SIP ET Da PortaSIP instance Envoi fea sip smartcall
84. ncepts Preface 2000 2009 PortaOne I nc All rights Reserved www portaone com This document provides PortaSIP PortaSwitch users with the most common examples and guidelines for setting up a VoIP network The last section of the document answers the most frequent questions users ask after running PortaSwitch for the first time Where to get the latest version of this guide The hard copy of this guide is updated at major releases only and does not always contain the latest material on enhancements occurring in between minor releases The online copy of this guide is always up to date integrating the latest changes to the product You can access the latest copy of this guide at www portaone com support documentation Conventions This publication uses the following conventions Commands and keywords are given in boldface Terminal sessions console screens or system file names are displayed in fixed width font Caution indicates that the described action might result in program malfunction or data loss NOTE Notes contain helpful suggestions about or references to materials not contained in this manual Timesaver means that you can save time by performing the action described in the paragraph Tips provide information that might help you solve a problem Porta SIP System Concepts Hardware and Software Requirements Server System Recommendations One UNIX Server A minimum of 80 GB of available d
85. ne payments A corresponding plugin can be developed for each new E911 provider so that you can effortlessly interconnect with them E911 address Since it is impossible to locate a customer s physical address using the IP address of his phone and asking the customer to provide his address during emergency calls is simply not acceptable every IP phone with a 911 service activated must have an address in the PSAP database before an actual emergency is ever made Therefore during registration the customer must provide an address where his device will be physically located and when he changes location e g goes on vacation he must update this address When a customer enters an emergency service address PortaBilling will validate it with the E911 provider to ensure that the address is valid and contains all the required information Then a link between phone number and address will be imported to the E911 provider database so that now if someone calls E911 from this phone the PSAP will receive complete information about the customer s location Special handling of 911 calls Of course PortaBilling applies a special policy for processing and routing emergency calls For instance even if a customer s account has exceeded its balance and he cannot make outgoing calls a 911 call will still go through Interconnection with an E911 provider Two steps are involved here e Connecting to the E911 provider s API to validate and popul
86. none se no yes w 216 231 44 1686 yes x 3600 no w HE HORDE HAE Supplementary Service Subscription yes Network Jitter Level SIP 100REL Enable Auth Resync Reboot MOH Server Use Outbound Proxy Use OB Proxy In Dialog Make Call Without Reg Ans Call Without Reg DNS SRY Auto Prefix User ID Use Auth ID Block CID Serv Dist Ring Serv Cfwd Busy Serv Cfwd Sel Serv Block Last Serv DND Serv CWCID Serv Call Back Serv Three Way Conf Serv Unattn Transfer Serv high v no v lt n Al gt 5 lt 1206001236 no M APPENDIX D Configuring Windows Messenger for Use as a SIP User Agent The following instructions apply to Windows Messenger version 5 0 1 Start Windows Messenger and select Options from the Tools menu 2000 2009 PortaOne I nc All rights Reserved www portaone com 96 Porta SIP Appendices t Windows Messenger Of x Options I want to 2 Check the My contacts include users of a SIP Communication Service check box Enter your Sign in name as shown in the form usernameQaddress where username is the name of the appropriate account in PB and address is either the IP address of the PortaSIP server or its name in DNS Then click the Advanced button 0118000 demo portaone com Signin name 2000 2009 PortaOne
87. number e To Called number condition This can be useful if you have multiple account aliases or DID numbers forwarded to your main account For instance you may wish to treat incoming calls to your business toll free number differently from calls to your regular phone number e Time Period Call time condition You can specify limitations regarding the time of day day of the week day of the month or some combination of these This is ideal for making sure your phone will not ring in the middle of the night A rule may contain only some of these limitations e g time in which case the others will contain a wildcard e g calls from any phone number ot made to any of your DID numbers Each rule provides instructions about exactly how a call should be processed It contains a sequence of one or more of the following actions e Reject Simply drop the call without answering it e Ring Ring on the current IP phone e Forward Redirect to the numbers defined in the call forward follow me settings e Voicemail Connect the call to this phone s voice mailbox When assigning an action to a tule you will be offered a list containing all the possible combinations based on the currently available features for this account For instance the Forward option will be present only if the call forwarding service is currently enabled for the account Call processing algorithm When a new call arrives to PortaSwitch call inf
88. omers to a vendor that only supports H 323 traffic what should do To do this you need to use a SIP gt H 323 protocol converter Either purchase a dedicated solution available from a number of vendors for instance Mera Networks www mera voip com or use one of your 36xx Cisco gateways with the special IOS feature called IPIPGW In addition to protocol conversion you may also need convert codecs This is not possible with IPIPGW but you can use the Cisco AS53XX gateway by looping one or more pairs of E1 T1 ports on it to allow SIP gt ISDN gt H323 call flow Please note that in the latter approach one ongoing session will consume 1 timeslot in each looped E1 T1 2 total as well as 2 DSPs For example if you have two El interfaces connected back to back the maximum number of simultaneous SIP sessions that you will be able to terminate to your H 323 provider will be 30 and each such session will use 2 DSPs have connected the Cisco AS53XX gateway to PSTN in order to send calls from PSTN to my SIP accounts and terminate calls from my SIP accounts to PSTN How many simultaneous sessions will it be able to handle A tule of thumb is that each SIP gt PSTN call or PSTN gt SIP call will use up one DSP and one timeslot in E1 T1 interface Therefore if you have connected your gateway to PSTN using for example two E1 ports and are using both of those ports for SIP lt gt PSTN the maximum number of simultaneous calls you will b
89. on a ptivate network with private IP address 192 168 0 7 NAT is the NAT server connected to the WAN via an interface with public IP address 9 8 7 6 and Server is the host on the WAN with which HOST 1 communicates Host 1 Server 5 i ps IP 192 168 0 7 Port 56789 0 IP 9 8 7 6 Port 12345 problem relating to the SIP User Agent UA arises when the UA is situated behind a NAT server When establishing a multimedia session the NAT server sends UDP information indicating which port it should use to send a media stream to the remote UA Since there is a NAT server between them the actual UDP port to which the remote UA should send 2000 2009 PortaOne Inc All rights Reserved www portaone com 33 Porta SIP System Concepts its RTP stream may differ from the port reported by the UA on a private LAN 12345 vs 56789 in the figure above and there is no reliable way for such a UA to discover this mapping However as was noted above the packets may not have an altered post translation port in all cases If the ports are equal a multimedia session will be established without difficulty Unfortunately there are no formal rules that can be applied to ensure correct operation but there are some factors which influence mapping The following are the major factors e How the NAT server is implemented internally Most NAT servers try to preserve the original source port when forwarding if possib
90. on according to the customer dialing rules or abbreviated dialing table so 3001234 will be converted into 12023001234 o Checks if A is actually allowed to call that number and what is the maximum allowed call duration o Discovers that the destination number is off net o Computes the routing for this call to the external vendors according to their cost and preferences and the customer s routing plan Based on the results of the above operations billing sends an authorization response to the SIP server 3 e The SIP server tries to send a call to all routes returned by the billing sequentially until either a connection is made or the list of routes is exhausted 4 e When the call is finished the SIP server sends accounting information to the billing 2000 2009 PortaOne Inc All rights Reserved www portaone com 15 Porta SIP System Concepts Terminating SIP calls to a vendor using VolP Porta M Billing SIP phone A Phone C e An example we are able to terminate calls to the US and Canada to a vendor X Telecom This would then be described as a VoIP to vendor connection in the billing with the remote address being the address of the vendor s SIP server or SIP enabled gateway e The billing engine returns the IP address of the vendor s SIP server in the route information with login password optional The PortaSIP server sends an INVITE request to that address providing the proper credentials and t
91. optional component used to ensure a proper media stream flow from one SIP telephone to another when one or both of them are behind a NAT firewall e Media Server The Media Server is used to play a number of short voice prompts to an SIP user when an error occurs such as zero balance invalid password and so on 2000 2009 PortaOne Inc All rights Reserved www portaone com 11 Porta SIP System Concepts Call Process Supported Services SIP UA lt gt SIP UA An example a customer purchases our VoIP services and two of his employees A and B are assigned SIP phone numbers 12027810003 and 12027810009 respectively For convenience the administrator creates two abbreviated dialing rules 120 for 12027810003 and 121 for 12027810009 Also he sets up standard US dialing rules so that users can dial local numbers in the 202 area code by just dialing a 7 digit phone number When the called party is online Porta M Billing SIP phone A SIP phone B This is the simplest case e User A dials user B s number 121 His SIP user agent sends an INVITE request to the SIP server 1 e The SIP server sends an authorization request to the billing 2 e Billing performs several operations o Checks that such an account exists that it is not blocked expired that the supplied password is correct that the account is allowed to use SIP services etc o Performs a dialed number translation according to the customer dialing
92. or Sipura to the same network as your SIP phone If possible disconnect the SIP phone and use the same IP address for the Cisco ATA Sipura as was previously used by the third party SIP phone Configure the Cisco ATA Sipura with the same account as was used on your third party SIP phone e Try to make test calls from the Cisco ATA Sipura e If you have followed the preceding steps and the problem disappears then this means your third party vendor SIP phone is not working according to the standard Contact the vendor of the SIP phone and describe the problem e If this problem with the Cisco ATA Sipura persists contact support portaone com Provide a full description of the 2000 2009 PortaOne Inc All rights Reserved www portaone com 84 Porta SIP Administration FAQ FAQ problem the ID of the account being used for testing and the relevant parts of the sip log and porta billing log Why can t my debit account initiate 3 way calling using the features of a SIP phone such as Cisco ATA 186 Since 3 way calling requires 2 simultaneous outgoing SIP sessions from one SIP telephone debit accounts will be unable to use it as the first session will lock the account and not allow the second one to go through Therefore if you want to enable your clients to use such services create a credit account for them instead Does PortaSIP support conferencing No Full scale SIP conferencing requires a separate software or hard
93. ormation is sequentially checked against all defined call processing rules The call information ANI DNIS and current time is checked against each rule s limitations If at least one of these does not match the rule is skipped and processing moves on to the next one If there is a match for all three limitations then the rule s actions are executed and no further rules are processed If none of the rules matches or if no call processing rules have been defined then the default rule is applied as follows e Ring on the IP phone 2000 2009 PortaOne Inc All rights Reserved www portaone com 54 Porta SIP System Concepts e If not answered within a certain time defined by the Timeout parameter in Service Features for the Voice Calls service and if the account has call forwarding enabled attempt to connect the call to the phone numbers listed there e If the call is still not answered and the account has the UM service enabled forward the call to voicemail otherwise drop the call Service Announcements via the Media Server A customer might be unable to make a call not only due to network problems but also for various administrative reasons for example if his account is blocked or he does not have enough money on his account If the end user can be informed of such administrative problems instead of just being given a busy signal this will greatly simplify troubleshooting Here is what would happen in the event th
94. r account A A gt B 15 minutes e Under account B A gt C 10 minutes As a result A does not really know that a call transfer took place A is charged for a normal outgoing call to B and this is what A will see in the CDR history B is charged for an outgoing call to C since B is responsible for the transfer 2000 2009 PortaOne Inc All rights Reserved www portaone com 47 Porta SIP System Concepts A scenario in which it is the calling party who initiates the transfer shown below is nearly identical to that described above for a transfer initiated by the called party Porta Billing PSTN GW O lt e lt lt Phone C 114 SIP phone A SIP phone B If A called B and after five minutes of conversation transferred B to C and they spoke for ten minutes there will be two CDRs both under account A e A gt B 15 minutes e B gt C 10 minutes Attended transfer Porta Billing SIP phone A SIP phone B 12 Phone C e A dials B s phone number 1 e PortaSIP sends the incoming call to B 2 when B answers the call is established between A and B 3 e B places A on hold 4 PortaSIP provides music on hold for A 5 e B initiates a new outgoing call to C 6 PortaSIP sends an authorization request to PortaBilling to check if B is allowed to 2000 2009 PortaOne Inc All rights Reserved www portaone com 48 Porta SIP System Concepts send a call to this destination and to ob
95. rmal PSTN gt SIP call the number is transformed the call is authorized in the billing 2 and the timer starts to measure the maximum call time allowed based on A s current balance 3 e The call is sent to PortaSIP 4 e PortaSIP receives the INVITE but without authorization information So the PortaSIP server performs authorization in the billing based on the IP address and also requests billing assisted routing 5 e PortaBilling recognizes that the destination is an account with follow me services enabled and produces a special list of routes o If the follow me mode chosen is When unavailable then a direct route to the account s SIP UA is included as the first route in the list with a default timeout o A list of follow me numbers is produced If the current time falls outside the specified period for a certain number it is removed from the list o If the follow me order is Random then the list of phone numbers is shuffled o The maximum call duration is calculated for each follow me number based on the balance and rates for the called account A o The resulting list of routes is produced and sent back to PortaSIP 6 2000 2009 PortaOne Inc All rights Reserved www portaone com 50 Porta SIP System Concepts e PortaSIP tries the first route 7 if the call is not connected within the timeout interval it moves to the next route 8 then to the next one 9 until either the call is put t
96. rmation according to what they know about themselves A client sitting behind a NAT knows only its internal IP port and this is what it enters in the SDP body of the outgoing SIP message When the destination endpoint wishes to begin sending packets to the originating endpoint it will use the received SDP information containing the internal IP port of the originating endpoint and so the packets will never arrive Understanding the SIP Server s Role in NAT Traversal Below is a simplified scheme of a typical SIP call SIP Server Livin mA g Media RTP UA 1 UA 2 It must be understood that SIP signaling messages between two endpoints always pass through a proxy server while media streams usually flow from one endpoint to another directly Since the SIP Server is located on a public network it can identify the real IP addresses of both parties and correct them in the SIP message if necessary before sending this message further Also the SIP Server can identify the real source ports from which SIP messages arrive and correct these as well This allows SIP signaling to 2000 2009 PortaOne Inc All rights Reserved www portaone com 35 Porta SIP System Concepts flow freely even if one or both UAs participating in a call are on private networks behind NATs Unfortunately due to the fact that an RTP media stream uses a different UDP port flowing not through the SIP server but directly from one UA to another there
97. roxy should be used to prevent no audio problems A call is made from one SIP UA SIP phone to another SIP UA SIP phone with both phones on the same private network behind the same NAT This scenario is likely to be encountered in a corporate environment where a hosted IP PBX service is provided In this case it is beneficial to enable both phones to communicate directly via their private IP addresses so that the voice traffic never leaves the LAN Calls between SIP phones and PSTN 1 A call is made from to a SIP phone on a public IP address from to a VoIP GW a VoIP GW is always assumed to be on a public IP address In this case the RTP stream may flow directly between the GW and SIP phone and no RTP proxying is required A call is made from to a UA under a NAT from to a VoIP GW and the remote gateway supports SIP COMEDIA extensions In this case the RTP stream may flow directly between the gateway and the SIP phone and there is no need to use an RTP proxy However you need to configure your Cisco GW as per APPENDIX B Cisco GW Setup for PortaSIP COMEDIA in order to ensure proper NAT traversal A call is made from to a UA under a NAT from to a VoIP GW and the remote gateway does not support SIP COMEDIA extensions An RTP proxy is required in this case 2000 2009 PortaOne Inc All rights Reserved www portaone com 37 Porta SIP System Concepts In appendices A through C you will find a list of teste
98. rver is a backend component that interacts with the PortaSIP proxy server and maintains online information for all users registered within your network It allows SIP user agents to publish subscribe requests and respond to them and to generate notifications of changes in presence status CO PortaSIP s presence service can run on the same physical server where the PortaSIP software package is installed Porta Billing e a RE UA T 8 o a Porta SIP S SIP UA f Proxy Presence T server server 5 SIP UA foo ooo SS Typically the whole process functions in the publish subscribe manner Presence information is published from a certain source e g mobile phones laptop computers PDAs desktop PCs or even other application servers The PortaSIP presence server then sends the combined presence data to all watchers who have subscribed to the presence service for the given user The presence server merges this information to form a complete overview of the each uset s presence information 2000 2009 PortaOne Inc All rights Reserved www portaone com 22 Porta SIP System Concepts Porta SIP TS Porta sip 7 server server Presence Watcher Presence Source e The SIP user agent sends a SUBSCRIBE request to the PortaSIP proxy server 1 if authorized successfully the SUBSCRIBE request is forwarded to the PortaSIP presence server
99. rvices a SIP phone must be able to automatically switch to backup servers should one of the SIP servers not be available How does a SIP phone know about alternative SIP servers There are several options 1 Program the backup SIP server s IP address into the SIP phones if this is supported by the IP phone configuration The main disadvantage of this method is that it only works with certain SIP phone models 2 Instead of programming the IP address of the SIP server into the SIP phone s config use a hostname instead e g sip supercall com This name can be set up to resolve to multiple IP addresses of different SIP servers DNS round robin However this may not work if the manufacturer of the SIP phone has employed a simplified approach so that the phone does not perform DNS resolving if a current SIP server fails 3 Use the DNS SRV records These records were designed specifically for the purpose of providing clients with information about available servers including the preferred order in which individual servers should be used in a redundant multi server environment This method is currently the most flexible and reliable one see details below Using DNS SRV records for multiple PortaSIP proxies an example Here we assume that you have two PortaSIP servers available in the main hosting center for your VoIP mysipcall com service as well as one backup PortaSIP server in a collocation center in a different city Your
100. s separated by a colon 6 A This file is created automatically during installation Thus assuming you provided correct parameters during installation you do not have to change anything General configuration Variable Description LADDR IP address of the SIP environment 2000 2009 PortaOne Inc All rights Reserved www portaone com 88 Porta SIP Administration FAQ Variable SIP_PORT CANONIC_NAME PB_MASTER PB_ROUTING_SER VER PB_REGISTER_SE RVER PB_ACCT_SERVER RAD_KEY AUTH_PORT ACCT_PORT RAD_TIMEOUT RAD_RETRIES Description Port on the SIP server which SIP phones should connect to value number default 5060 Fully qualified domain name for this SIP server so your customers can use contact information in the form 1234 sip domain com PortaBilling virtual environment id for this SIP instance note that this is a numeric ID i_env and not the environment name use the porta admin pl utility on the slave server to find the correct value RADIUS configuration IP address of the PortaBilling100 master host IP address of the PortaBilling100 RADIUS server used to process authorization routing requests if different from the PB_MASTER above IP address of the PortaBilling100 RADIUS server used to process registration requests if different from the PB_MASTER above IP address of the PortaBilling100 RADIUS server used to process accounting requests if different from the PB_M
101. served www portaone com 90 Porta SIP Administration FAQ Starting Stopping PortaSIP Services If you need to stop all PortaSIP services then execute the following command sudo usr local etc rc d sip sh stop This will properly terminate all components To start PortaSIP use the following command 5 sudo usr local etc rc d sip sh start NOTE Please always make sure that you have stopped services as described above before trying to start them again since trying to start services when they are already running may render the service inoperable 2000 2009 PortaOne Inc All rights Reserved www portaone com 91 Porta SIP Appendices 6 Appendices 2000 2009 PortaOne Inc All rights Reserved www portaone com 92 Porta SIP Appendices APPENDIX A Supported SIP RFCs RFC 3261 SIP Session Initiation Protocol supported with the limitation that the SIP URL domain is ignored in the incoming requests RFC 4566 SDP Session Description Protocol RFC 2327 SDP Session Description Protocol supported with the limitations and relaxations provided for SDP under SIP RFC 3263 Session Initiation Protocol SIP Locating SIP Servers supported RFC 3264 An Offer Answer Model with the Session Description Protocol SDP only the early model is supported RFC 3265 Session Initiation Protocol SIP Specific Event Notification supported in the presence ser
102. starts immediately with Forward Otherwise the system will first ring only your IP phone and then ring both your IP phone and all the other phones SIP URI forwarding In traditional call forwarding you only specify a phone number where calls are sent using the currently available termination partners This is very convenient for calls terminated to PSTN since in this case PortaSwitch LCR profit guarantee fail over and other routing capabilities are engaged automatically If you provide services such as DID exchange however calls must be forwarded directly to a large number of different SIP proxies belonging to your customers In this case for every account DID you simply define which phone number and IP address all incoming calls should be forwarded to In order to protect you from abuse of this service e g a customer tries to set up call forwarding to somebody else s network then relays a storm of 2000 2009 PortaOne Inc All rights Reserved www portaone com 51 Porta SIP System Concepts call attempts through your SIP server it is only possible to use those SIP proxies which are listed in the Permitted SIP Proxies customer information If a customer who buys DIDs from you has two SIP proxies you need two list each of those proxies in the Permitted SIP Proxies configuration After that your administrators or the customer on his self care pages will be allowed to use these IPs in the SIP URI Billing forwarded
103. t to contact them in sequence until it succeeds Remote Party ID lt sip 1234 sip example com gt party callid privacy full Understanding SIP Call Routing When the PortaSIP server has to establish an outgoing call it must find out where the call is being sent to To do this it will ask billing for a list of possible routes In this case the routing configuration is in one central location and billing can use information about termination costs quality or other parameters to choose the best route least cost routing quality based routing profit guarantee individual routing plans etc When a call goes through the PortaSIP server the SIP server may e Direct the call to one of the registered SIP clients if the called number belongs to the registered agent e Optionally direct the call to the voicemail box PortaUM required if the called number belongs to an account in PortaBilling but this account is not currently registered to the SIP server is offline e Route the call to one of the gateways for termination according to the routing rules specified in PortaBilling Please consult PortaBilling Administrator Guide for more information about various routing parameters and methods 2000 2009 PortaOne Inc All rights Reserved www portaone com 32 Porta SIP System Concepts NAT Traversal Guidelines NAT Overview The purpose of NAT Network Address Translation is to allow multiple hosts on a private LAN not dire
104. ta sip conf chapter for details on RTP proxy policy configuration Auto provisioning IP Phones If you provide your VoIP customers with IP phone equipment you know how laborious and yet important the task of performing initial configuration is If the equipment is not configured properly it will not work after being delivered to the customer Or even if it works initially problems will arise if you need to change the IP address of the SIP server How can you reconfigure thousands of devices that are already on the customer s premises There are two ways to manage the device configuration Manual provisioning The administrator must login to the device provisioning interface typically HTTP and change the required parameters There are several drawbacks to this method e The IP phone must be connected to the Internet when the administrator is performing this operation e The administrator must know the device s IP address e The IP phone must be on the same LAN as the administrator or on a public IP address if the device is behind a NAT firewall the administrator will not be able to access it 2000 2009 PortaOne Inc All rights Reserved www portaone com 40 Porta SIP System Concepts Due to these reasons and since every device must be provisioned individually this method is acceptable for a testing environment or small scale service deployment but totally inappropriate for ITSPs with thousands of IP phones around th
105. tain the routing 7 In the case of a positive reply PortaSIP establishes a call to C 8 The call is now established between B and C 9 after a short exchange B decides to bridge A and C together and a REFER message is sent to PortaSIP 10 PortaSIP will now connect A and C together 12 and cancel both of the call legs going to B When either A or C hangs up the call is terminated and two accounting records are sent to the billing 13 one is for the A gt B call charged to its originator A and the other for the A gt C call likewise charged to its originator B Call Forwarding PortaSIP supports several call forwarding modes you can select a specific mode from the Forward Mode menu on the Call Features tab Forward to CLD is simple unconditional forwarding to a different phone number Follow me allows you to specify multiple destinations for call forwarding each of which is active in its own time period You can also specify that multiple numbers be tried one after another or that they all ring at the same time Forward to SIP URI allows you to specify not only a destination phone number but also an IP address for calls to be forwarded to This is useful when calls have to be routed directly to an external SIP proxy Advanced Forwarding adds a few extra options to those available in Follow me mode and also allows you to route calls to SIP URI It thus represents a super set of all call forwarding capabilities Fo
106. tion capability e g Cisco AS5350 78 Make all SIP calls to a certain prefix NNN go to my gateway XXX 78 allow my customer to have two phone numbers from different countries which will both ring on the same SIP phone 79 create an application to handle PSTN gt SIP calls 79 configure SIP phone X made by vendor Y 79 bill incoming calls from PSTN to SIP using a special rate 80 bill using different rate plans for incoming outgoing and forwarded GANS 2 Ee E E E EE 80 provide error messages from the media server in my users local language 81 calculate how much bandwidth need for my PortaSIP server 81 enable my SIP phone or ATA to be automatically provisioned by PORES WIC M2 cs seen none ei 82 Administration FAQ 83 Troubleshooting Common Problems 84 ET tate 85 PortaSIP Configuration 88 POI n E 92 APPENDIX A Supported SIP RFCS 93 APPENDIX B Cisco GW Setup for PortaSIP COMEDIA eee 93 APPENDIX C Client s Sipura Configuration for PortaSl P ow eee 94 APPENDIX D Configuring Windows Messenger for Use as a SIP User QIN E E A EE E sn 96 APPENDIX E SJ Phone Configuration for PortaSIP 99 APPENDIX F SIP Devices with Auto provisioning 101 Porta SIP System Co
107. to PortaSIP server 1 1 e The SIP server sends an authorization request to the billing 2 e After all the usual authorization checks the billing discovers that the dialed number is one of our SIP accounts but is currently registered to PortaSIP server 2 It instructs the SIP server to route this call to the IP address of PortaSIP 2 3 e PortaSIP server 1 sends an INVITE request to PortaSIP server 2 4 e Upon receiving this INVITE PortaSIP 2 sends an authorization request to the billing 5 e The billing authorizes the call since it comes from a trusted node and requests that the call be sent to the locally registered SIP UA 6 e The SIP server sends an INVITE request to the SIP phone 7 2000 2009 PortaOne Inc All rights Reserved www portaone com 14 Porta SIP System Concepts SIP UA gt PSTN Porta M Billing SIP phone A GW NY 02 Phone C 12 34 56 78 e User A attempts to call his co worker user C C has not been assigned a SIP phone yet thus he only has a normal PSTN phone number from the 202 area code and A dials 3001234 As SIP user agent sends an INVITE request to the SIP server 1 e The SIP server sends an authorization request to the billing 2 e Billing performs several operations o Checks that such an account exists that it is not blocked expired that the supplied password is correct that the account is allowed to use SIP services etc o Performs a dialed number translati
108. to send the call to A s SIP phone 12027810003 4 e PortaSIP sends a call setup request to the SIP phone 5 e If the dialed number belongs to a SIP account with unified messaging services enabled and the account is not online at the moment or does not answer the call will be redirected to a voicemail system After the call is completed A is charged for it also costs are calculated for the incoming call according to the tariff associated with X Telecom s VoIP from Vendor connection 2000 2009 PortaOne Inc All rights Reserved www portaone com 20 Porta SIP System Concepts Separate RTP Proxy Server In the normal scenario all PortaSIP components reside on the same physical server so both SIP signaling and RTP media pass through it Although PortaSIP does not perform transcoding of voice traffic the many concurrent calls passing through the server still put a certain load on the system as a result of the huge number of relatively small packets that need to be processed Porta SIP s SIP SIP Signaling Signaling 9 7 q Ree Re Voice traffic is quite different from other types of Internet traffic e g web downloads as it is very sensitive to packet delays At any given moment the system needs to transport packets at a constant speed and with minimum added delay If a PortaSIP server is loaded with other tasks this may become difficult So now you have the option of installing
109. tomer A tries to call B and the call is connected e While the call is in progress PortaSIP periodically sends a small SIP request to the SIP phone e If the phone replies this means that the phone is still online e If no reply is received PortaSIP will attempt to resend the keep alive packet several times this is done to prevent call disconnection in the case of an only temporary network connectivity problem on the SIP phone side e If no reply has been received following all attempts PortaSIP will conclude that the SIP phone has unexpectedly gone offline and will disconnect the other call leg and send an accounting record to the billing e Therefore the call will be charged for call duration quite close to the real one First Login Greeting This feature is not directly related to call processing but will give your PortaSwitch based VoIP service a competitive advantage When a customer unpacks his new SIP phone and connects it to the Internet the phone will start ringing When the customer picks up the phone he will hear a greeting recorded by you congratulating him on successfully activating his VoIP service and giving him other important information If the customer does not answer the phone e g he has connected his SIP adaptor to the Internet but has not connected the phone to it yet and so cannot hear it ringing PortaSIP will try to call him back later Of course after the customer has listened to the message once
110. urrently is the same as adding flavorings salt pepper etc to a stew by following several recipes from different cookbooks at the same time even a slight mix up will probably result in your adding some of the seasonings twice while not putting others in at all and the result will be something which no one can eat Currently one very common problem situation is that where a SIP phone is behind a symmetric NAT and obtains its public IP address from STUN putting this into the contact information This confuses the RTP proxy since PortaSIP regards the SIP phone as being on a public IP address so that no RTP proxy is used the result is one way audio So the simplest answer is yes You can use STUN to avoid usage of an RTP proxy in some cases At the present moment however due to unreliable STUN support on the IP phone side the safest option is to avoid using STUN PortaSIP Configuration PortaSIP provides a unified configuration tool Even if a system consists of several components using different technologies and configuration methods you just have to edit one simple configuration file This master configuration file is then used by PortaOne configuration scripts to manage and provision other modules e g SIP Proxy B2BUA and so on porta sip conf This is the only file you need to edit in order to modify PortaSIP parameters Every row starting with is considered to be a comment the other lines will contain VAR VALUE pair
111. user agent cannot register with the SIP server and report I am going to receive all calls for prefix NNN Cisco 5300 supports the REGISTER command but this only works for numbers assigned to FXS ports or IP phones Therefore if you have a gateway with E1 T1 connected to it and wish to route certain prefixes there for termination you must define the routing in the billing To do this proceed as follows e Create a new tariff with the Routing Ext e When you enter rates into this tariff two new columns will appear Preference and Huntstop Enter the desired routing 2000 2009 PortaOne Inc All rights Reserved www portaone com 78 Porta SIP How to preference The higher the number the more desirable this route is 0 means no route at all Turn the huntstop on if you do not wish to use any routes with a lower priority e Create a PSTN to vendor connection to the vendor specify the gateway which will handle termination as your Node and select the tariff you have created as the termination tariff e Make sure that your gateway is actually configured to accept incoming VoIP calls and send them to telephony for the destinations you plan to terminate allow my customer to have two phone numbers from different countries which will both ring on the same SIP phone You can have an unlimited number of such extra phone numbers Your customer will have one main account e g 12027810003 which will be provis
112. users normally use either one of the main servers and only if they cannot access either of them e g a network problem affecting the entire hosting center will they go to a backup one First of all your DNS server for the mysipcall com domain must be configured with DNS A records for the individual PortaSIP servers portasipl IN A 193 100 3522 portasip2 IN A 193 100 3 5 portasip3 IN A 64 12 63 37 After this you may define a SRV record describing the available SIP servers _sip _udp proxy SRV 10 0 5060 portasipl SRV 10 0 5060 portasip2 SRV 60 0 5060 portasip3 2000 2009 PortaOne Inc All rights Reserved www portaone com 31 Porta SIP System Concepts The first two servers have a higher priority 10 so they will be tried first Also note that DNS SVR allows you to specify which port should be used for communication On your SIP phone you should specify the following SIP proxy registrar proxy mysipcall com Use DNS SRV yes DNS SRV Auto Prefix yes If you do not switch on the auto prefix feature then the SIP proxy address must be entered as _sip _udp proxy mysipcall com So now when a SIP phone is switched on it will first query the DNS database for servers for _sip_udp_ proxy mysipcall com receiving a list of recommended servers portasip1 mysipcall com portasip2 mysipcall com and portasip3 mysipcall com After that it will obtain the IP addresses of these servers from the DNS database and attemp
113. ussed earlier in the SIP UA lt gt SIP UA section e PortaSIP receives an incoming call and requests authorization and routing from PortaBilling100 e PortaBilling verifies whether this call should be allowed and if the destination is one of our SIP accounts e PortaBilling checks the registration database and returns the address of the PortaSIP server the account is currently registered to in the routing information e PortaSIP receives its own address as the route and sends a call to the SIP phone 2000 2009 PortaOne Inc All rights Reserved www portaone com 28 Porta SIP System Concepts Case B SIP phones registered to different PortaSIP servers In this case routing information from PortaBilling will contain the address of the second PortaSIP server i e the one to which the called SIP phone is registered Thus the first PortaSIP server will send a call there and then the second PortaSIP server will send the call to the SIP phone Porta K Billing Billing Engine En E a gt PortaSIP PortaSIP It may be asked why the first originating PortaSIP server does not send the call directly to the called SIP phone since the registration database contains its contact IP port information The answer is that if the called SIP phone is behind a NAT and most Internet users are behind a NAT these days only the server on which the SIP phone has opened a connection can send back a reply and t
114. vendor and set up the NAT traversal status accordingly of N nq NAT ZI traversal call 1 is SN A Porta KASIP Siy ISS 5 NAT traversal 7 NAT available call 2 RTP ji VendorB SS ye 7 2000 2009 PortaOne Inc All rights Reserved www portaone com 39 Porta SIP System Concepts After the NAT status of the IP phone behind NAT or on a public IP and the NAT traversal status of the connection have been identified a decision is made as follows e Ifthe connection has Always NAT traversal status activate the RTP proxy e Ifthe connection has Direct NAT traversal status do not activate the RTP proxy e If the phone is behind NAT and the connection has OnNat status activate the RTP proxy e Otherwise do not activate the RTP proxy In addition to the option of media proxying based on a specific vendor s proxying policy it is also possible to activate full media proxying for a specific account phone line or a specific customer all accounts under the customer This can be used to force NAT traversal on the PortaSwitch side in complex network configurations or to provide users with an extra level of privacy All of this is related to the smart logic of RTP proxying Of course you have control over the RTP proxy s behavior and may change the default policy for instance you may permanently switch the RTP proxy off See por
115. ver and emulated in the B2BUA RFC 3323 A Privacy Mechanism for the Session Initiation Protocol SIP supported in part RFC 3324 Short Term Requirements for Network Asserted Identity 3325 Private Extensions to the Session Initiation Protocol SIP for Asserted Identity within Trusted Networks supported for outgoing call legs only RFC 3428 Session Initiation Protocol SIP Extension for Instant Messaging supported RFC 3489 STUN Simple Traversal of User Datagram Protocol UDP Through Network Address Translators NATs supported RFC 3515 The Session Initiation Protocol SIP Refer Method supported RFC 3581 An Extension to the Session Initiation Protocol SIP for Symmetric Response Routing supported RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol SIP supported APPENDIX B Cisco GW Setup for PortaSIP COMEDIA sip ua nat symmetric check media src 2000 2009 PortaOne Inc All rights Reserved www portaone com 93 Porta SIP Appendices APPENDIX C Client s Sipura Configuration for PortaSIP 1 First you need to know the SPA IP address Via a touchtone telephone attached to the phone port on the SPA press the stat key four times Then type 110 and the IP address will be announced 2 Runa Web browser application on the same network as the SPA Open a s
116. ware solution However you can make use of the features available in some SIP phones such as Cisco ATA 186 to allow your clients to set up simple so called chain conferences For more information please refer to the documentation for each specific SIP phone Can you assist me in integrating SIP device X gateway media server conference server etc made by vendor Y with PortaSIP Yes we can however you will have to purchase an additional consulting contract Generally speaking there should be no compatibility problems between PortaSIP and any standards compliant SIP device However for obvious reasons we only provide detailed setup instructions for the Cisco AS5300 gateway Can I use PortaSIP with a billing system other than PortaBilling100 Yes this is possible PortaSIP uses the standard Radius protocol to communicate with the billing engine and its AAA behavior was purposely made very similar to that of Cisco IOS So it should work with any billing system that supports Radius and can bill Cisco gateways However advanced services such as billing assisted routing abbreviated dialing PortaUM integration and so on require support from the billing engine Detailed specifications of the protocol used to exchange information between PortaBilling100 and PortaSIP are available upon request 2000 2009 PortaOne Inc All rights Reserved www portaone com 85 Porta SIP Administration FAQ I want to terminate my SIP cust
117. where you provide services You can activate the Legal Intercept call feature in PortaBilling for every account that requires it obviously this feature is only accessible from the administrator interface and is not visible to the end user When this is done PortaSIP will be instructed to engage the RTP proxy for every outgoing or incoming call to this account regardless of other NAT traversal settings and will produce a complete call recording of the conversation The call recordings may then be delivered to the law enforcement agency by any applicable means or you may even provide real time access to the location on the PortaSIP server where these files are stored In the specific case of CALEA there are many requirements which an ITSP must comply with many of them not even related to technical capabilities but rather purely to administration e g personnel dealing with intercept data must have an appropriate security clearance So the optimal solution for ITSPs using PortaSwitch is another option described by CALEA i e going via a trusted third party At present PortaSwitch has been successfully tested with the Just in Time product from NeuStar s Fiduciary Services 2000 2009 PortaOne Inc All rights Reserved www portaone com 64 Porta SIP System Concepts Secure Calling PortaSIP fully supports Secure Real time Transport Protocol SRTP according to RFC 3711 which provides confidentiality message authent
118. y Cisco gateway to send outgoing calls using SIP Configuration of the Cisco gateway for SIP is not much more difficult than H323 First of all make sure that the rest of your system is configured properly that the gateway can place the outgoing calls and is able to communicate with the billing using RADIUS SIP server parameters Specify general parameters of the SIP server such as hostname You can also refer to the SIP server by its IP address however this method will require reconfiguration of each individual gateway if you change the IP address of your SIP server sip ua aaa username proxy auth sip server dns lt hostname of your SIP server gt NOTE Cisco GWs are currently unable to register to SIP servers using the REGISTER method or to perform proper authorization of an outgoing call using the INVITE method Therefore remote IP address authorization is performed by PortaSIP when it detects an incoming call from the Cisco gateway In order for this authorization to be successful the gateway should be registered among the PortaBilling nodes Dial peers Now you can create an SIP enabled outgoing dial peer dial peer voice 200 voip destination pattern T session protocol sipv2 2000 2009 PortaOne Inc All rights Reserved www portaone com 76 Porta SIP How to session target sip server You probably will need an application on the incoming telephony dial peer to properly authenticate and authorize in
119. ystem Concepts Clustering of PortaSIP Servers You may also install several physically independent PortaSIP servers and connect all of them to the same virtual environment in PortaBilling100 In this case several PortaSIP servers combined in this case into a PortaSIP cluster communicate with a single central billing which provides all the required service provisioning information and maintains a global database of SIP phone registrations A SIP phone user may connect to any of the available PortaSIP servers only those which are available to him via his product s accessibility of course Once a SIP phone is successfully registered to one of the SIP servers the information is globally available within this PortaSwitch environment Porta K Billing f Billing Engine Billing SIP Provisioning Registrations PortaSIP PortaSIP K re By installing several independent PortaSIP servers you can achieve two main goals e Improve the reliability of your network e Optimize call flow on your network so as to better utilize the available network infrastructure 2000 2009 PortaOne Inc All rights Reserved www portaone com 26 Porta SIP System Concepts Improved Reliability Porta K Billing Billing Engine Billing SIP Provisioning Registrations A PortaSIP PortaSIP PortaSIP Even if one of the SIP servers is down due to network issues or hardware problems your subscribers can continue using t

Download Pdf Manuals

Related Search

Related Contents

Philips Blu-ray Disc/ DVD player BDP2930  USER`S MANUAL CCTV TesterPro  Guide de démarrage des logiciels  Philips 37PFL9903 Flat Panel Television User Manual  ISL29035EVAL1Z User Guide  programme - port  

Copyright © All rights reserved. Failed to retrieve file