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PortaBilling: User Manual

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1. SOS Porta X4Billing Porta K SIP PortaSIP instanc customers sip smartcall 195 70 140 Environment A PortaSIP instan sip supercall 195 70 14 tomers env B Environment B Every virtual SIP server acts as an independent PortaSIP installation 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 2000 2006 PortaOne Inc All rights reserved www portaone com 21 Porta SIP System 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 th
2. 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 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
3. 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 H 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 2006 PortaOne Inc All rights reserved www portaone com 74 Porta SIP Administration FAQ 3 Administration FAQ 2000 2006 PortaOne Inc All rights reserved www portaone com 15 Porta SIP 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
4. has been defined and is now applied to the phone number with the result 01142021234567 Note that there may be several carriers who can terminate this call each with its own numbering format In such a case there will be several alternative routes with different phone numbers 4 PortaSIP attempts to establish a connection to remote gateway 1 2 3 4 using phone number 01142021234567 5 After the call is completed PortaSIP sends an accounting request to PortaBilling stating that a call to remote gateway 1 2 3 4 has just been completed PortaBilling finds a connection to vendor ABC with remote IP address 1 2 3 4 and applies the translation 2000 2006 PortaOne Inc All rights reserved www portaone com 47 Porta SIP System Concepts rule s 011 for this connection in order to convert the number from the vendor specific format into your billing format Thus 011 is removed from 01142021234567 and the number becomes 42021234567 PortaBilling searches for the vendor and customer rates for this number and produces the CDRs CLI translation rules off net calls CLI ANI is the calling party number typically programmed on SIP phones However due to the reasons described above this number must be represented in a specific format depending on the situation For instance when your SIP account 12027810003 makes an off net call to the United States PSTN network the ANI number must be in the 10 digit format area code phone numbe
5. l l l 10 19 48 gt gt I 10 19 48 lt w an m Unauthor l 10 19 48 gt Rx I l 10 19 48 gt a 102 1 gt INVITE gt I 10 19 48 lt 102 T 10 ing I 10 19 48 gt a 102 1 INVITE gt 10 19 48 lt 102 100 Trying 10 19 48 l G gt Authorization request gt 10 19 48 I lt Auth reouest accented f 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 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 cas
6. summatized 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 e Otherwise the RTP proxy is not 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 2000 2006 PortaOne Inc All rights reserved www portaone com 55 Porta SIP System Concepts poor connection to the rest of the world Thus you will need to proxy his traffic to e
7. 0 0 AltGKTimeOut 0 GkTimeToLive 300 2000 2006 PortaOne Inc All rights reserved www portaone com 87 Porta SIP Appendices GkId i seSIP 1 SIPRegInterval 180 MaxRedirect 5 SIPRegOn 1 NATIP 0 0 0 0 SIPPort 5060 MediaPort DIFFERENT FOR EACH CLIENT AS DESCRIBED IN THE SETUP GUIDELINES OutBoundProxy 0 0 0 0 NatServer IP ADDRESS OF SERVER RUNNING PORTASIP NatTimer Oxle LBRCodec 0 AudioMode 0x00150015 RxCodec 0 TxCodec 0 NumTxFrames 1 CallFeatures Ox ffffffff PaidFeatures xffffffff CallerIdMethod xc 019e60 FeatureTimer 0 Polarity 0 ConnectMode 0xe0400 AuthMethod 0 TimeZone SEE CISCO ATA 186 DOCUMENTATION FOR ENTERING CORRECT ALUE NTPIP 192 43 244 18 AltNTPIP 131 188 3 222 DNS1IP 0 0 0 0 DNS2IP 0 0 0 0 UDPTOS Oxb8 SigTimer 0x64 OpFlags 0x62 ANSettings Ox2b NPrint 0 0 0 0 TraceFlags 0 The manufacturer s default values are assumed for all options not listed here 2000 2006 PortaOne Inc All rights reserved www portaone com 88 Porta SIP Appendices APPENDIX D 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 star 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 session in the SPA by typing http lt spa ip address gt admin advanced 3 Choose th
8. 4 2 5 4 1 5 2 Make sure you count the RJ 48C pins as shown in the illustration below fe i ge i 12345678 12345678 Hook underneath Hook underneath quun ABAALAA EEE SS Ss aA ZA AA gt A N N N N N N N N N N PRI T1 E1 CrossOver Loopback Cable Alternatively you can order ready made ones You can find a number of vendors producing such cables by searching for RJ 48C cross over cable on www google com Once the cable is ready plug it into the designated pair of T1 E1 ports in your Cisco AS5300 gateway 2000 2006 PortaOne Inc All rights reserved www portaone com 96 Porta SIP Appendices Software Configuration You also have to configure the T1 E1 interfaces The sample configuration below is for T1 adjust the time slots for El isdn switch type primary 5ess I controller T1 0 framing sf clock source line primary linecode ami pri group timeslots 1 24 controller T1 1 framing sf clock source line secondary 1 linecode ami pri group timeslots 1 24 I controller T1 2 framing sf linecode ami pri group timeslots 1 24 controller T1 3 framing sf linecode ami pri group timeslots 1 24 1 interface Serial0 23 no ip address isdn switch type primary 5ess isdn protocol emulate network no cdp enable interface Seriall 23 no ip address isdn switch type primary 5ess no cdp enable I interface Serial2 23 no ip addre
9. 44 168 CPE CET ET ETES Supplementary Service Subscription yes M yes M yes M 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 BS yes no M yes w no M no M no Vv 1206001236 no w APPENDIX E 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 2006 PortaOne Inc All rights reserved www portaone com 91 Porta SIP Appendices Windows Messenger Iof 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 2000 2
10. All rights reserved www portaone com 3 4 Porta SIP System Concepts PortaSiP controlled call transfer Typically in a 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 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 activities e g when an unsuspecting victim is transferred to a very expensive international desti
11. USD Default 5 ny gw 01 VendorB Termination to carrier B VendorB N 2 8610 CHINA Beijing 0 04000 USD Cheap 7 N ny gw 01 VendorA Termination to carrier A Vendor A 3 86 CHINA Proper 0 02500 USD Cheap 6 N 45 12 156 200 Vendor D Termination to vendor D Vendor D 4 86 CHINA Proper 0 03000 USD Cheap 6 N 5 86 CHINA Proper 0 11000USD Expensive 5 N ny gw 01 VendorC Termination to carrier Vendor C 193 50 123 6 VendorE Termination to carrier E Vendor E Fail over routing If a route fails e g the remote gateway is not available or could not complete the call because there was no available telephony port or therewas another problem on the vendor s side PortaSIP will automatically try to deliver the call via the next route If that route fails as well PortaSIP will try the one following it and will keep trying until either the call is connected or there are no more routes left Number translation There are many different phone number formats some used by your customers others by your vendors How to deal with all of them without making mistakes PortaBilling offers a powerful tool called translation rules for converting phone numbers with several different types depending on customers needs Your network numbering plan The key to avoiding problems with number formats is to choose a certain number format as the standard for your network and make sure that calls travel on your network only in this format The ideal candidate
12. 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 2006 PortaOne Inc All rights reserved www portaone com 11 Porta SIP Administration FAQ I want to terminate my SIP customers 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
13. a copy All product names mentioned in this manual are for identification purposes only and are either trademarks or registered trademarks of their respective owners 2000 2006 PortaOne Inc All rights reserved www portaone com 1 Porta SIP PortaSIP Administrator Guide Table of Contents PROT ACG sccccecsscisdezssesvssssussetieteassaaspanisdictsaaezssise ealettadsed antayisteaielastvaaatteaeenatteadstasastaaieaatiedt 4 Hardware and Software Requirements c cccscsecssesssssssesseesseessessesseesseesesseess 5 Let SHAN AGO Wi E ane 6 What s New in Maintenance Release 137 6 Important upgrade notes 7 1 SYSEMCONCOD S emmener 8 PortaSIP s Role in Your VolP Network 9 PortaSIP Components 11 Call Process Supported Services 12 Virtual SIP Servers 21 Clustering of PortaS P servers 22 Call flow scenarios for a PortaS P cluster 24 Advanced Features ceccecssecssesssesssessessecssecsuessecssscsuesssecsesssecssecssecsscssseaseeseeeseeeseess 28 Understanding SIP Call Routing 40 NAT Traversal Guidelines oo ccccccsccssssssscssecssecssscscssscsssessesssssseessecssessesseesnecsee 49 Auto provisioning of IP phones 57 PortaSIP and E911 services 59 IP Centrex features 61 2 HOW Cae ase aaa cca EEE Eee Rene sec see E 66 configure my Cisco gateway to accept incoming SIP calls and termin
14. calls outgoing calls toll restrictions code restrictions and differential treatment for internal and external calls Provided using the tariff configuration in PortaBilling 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 1 Some issues have been detected with Sipura ATAs working with the call park IVR this issue should be resolved in the new versions of Sipura firmware please contact PortaOne support for more details 2000 2006 PortaOne Inc All rights reserved www portaone com 62 Porta SIP System Concepts 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 Alerts the user to incoming calis when the user s line is busy with an established call Upon hearing the Call Waiting tone the user can put the current conversation on bold to answer the incoming call Supported by PortaSwitch assuming that the Call Waiting service has been enabled on the IP phone 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 cal
15. 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 E1 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 In APPENDIX F S Phone Configuration for PortaSIP 7 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 W Service PortaOne Please enter this information to initialize the service profile Account 123456789 Password ss ele X Save service information permanently 2000 2006 PortaOne Inc All rights reserved www portaone com 18 Porta SIP Administration FAQ 2 Key in the Account ID and password for the PortaSIP and press OK SJPhone 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 2006 PortaOne Inc All
16. dial peer to properly authenticate and authorize incoming 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 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 matc
17. for such a format is E 164 of course it is highly recommended that you use this same format in billing as well When a call arrives from your customer with a phone number from a customer specific number plan PortaSwitch will convert the number into your network format It will then travel on your network until it is sent to a vendor for termination Just before this happens it can be converted into the vendor specific format 2000 2006 PortaOne Inc All rights reserved www portaone com 45 Porta SIP System Concepts Customer based translation rules Very often your customer will have his own numbering format for example dialing 00 for international numbers while dialing just the phone number for local calls Customer based translation rules allow you to convert a number from a format specific to this particular customer There is a special dialing rules wizard available to make such configuration easier so that customers can even do this themselves Customer based translation rules have two applications e When a number is submitted for authorization these rules will be applied with the resulting number used to search rates Thus if your customer dials 0042021234567 you can convert it to 42021234567 and find the correct rates for the 420 prefix e This number will be returned to the node which requested it Connection based outgoing translation rules If your vendor requires a special number format e g tech prefix
18. 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 2 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 proxy should be used to prevent no audio problems 2000 2006 PortaOne Inc All rights reserved www portaone com 53 Porta SIP System Concepts 3 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 2 A callis 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
19. gateway will receive approximately the same number of calls 33 e A list of these IP addresses with optional login and password for SIP authentication will be returned to the SIP server To avoid extremely long delays only a certain number of routes from the 2000 2006 PortaOne Inc All rights reserved www portaone com 42 Porta SIP System Concepts beginning of the list are returned the default is 15 but this can be changed in porta billing conf Route sorting How exactly does PortaBilling100 arrange multiple available routes 1 By route category Only route categories which are included in the routing plan will be used following the order given in the routing plan 2 Ifyou have multiple route categories within the routing plan you can either merge them into the same group by assigning them the same order value or keep each one separate with its own order value Then routes within the same order group for route categories will be arranged according to preference 3 For routes with the same preference the system can arrange them according to cost a comparison is made on the Price_Next rate parameter so that cheaper routes will be among the first ones or in random fashion Does PortaSwitch support LCR Yes we support LCR and much more besides In fact just LCR is the simplest type of routing PortaSwitch handles If you decide not to use routing plans one default plan for everyone or routing pre
20. in the default configuration 2000 2006 PortaOne Inc All rights reserved www portaone com 6 Porta SIP PortaSIP Administrator Guide Important upgrade notes We try to make the process of upgrading as easy as possible and to keep our releases backward compatible There are just a few things you should pay attention to when upgrading e PortaSIP Maintenance Release 13 is supplied with FreeBSD 6 1 with enhanced reliability and performance capabilities If you are currently using PortaSIP with FreeBSD 5 x we strongly encourage you to backup your PortaSIP configuration and perform a full reinstallment from the provided PortaSIP CD It will install FreeBSD version 6 1 as well as PortaSIP Maintenance Release 13 2000 2006 PortaOne Inc All rights reserved www portaone com 7 Porta SIP System Concepts 1 System Concepts 2000 2006 PortaOne Inc All rights reserved www portaone com 8 Porta SIP System Concepts PortaSIP s Role in Your VolP Network i Billing Engine ATA186 Router A x ga Portal 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 lo
21. is behind a NAT bill SIP to SIP calls By default calls from one SIP account to another are treated as on net ones and are therefore not billed However if you want to bill your customers for such calls you can do the following e Add the appropriate rate to the tariff associated with the accounts to be charged For example if you have SIP accounts with the prefix 078 then you should add the appropriate rate for destination 078 to the tariff used to charge for outgoing calls e Create a special tariff with rates corresponding to the prefixes allocated for your SIP accounts 078 in the example above This will be the tariff used to calculate your termination expenses Since you do not pay anything for such termination you can enter zero prices for all of the rates e Create a new vendor with a descriptive name for example Direct termination to SIP phones Add a VoIP to Vendor connection to that vendor with the tariff created in the previous step and enter sip ua in the Remote IP field So now if a call is made from one SIP phone to another the originating party will be charged according to the rates you have entered in the customer s tariff This call will be counted as terminated to the vendor Direct termination to SIP phones with zero termination cost but it will still be recorded in the database so you can easily view statistics for all SIP SIP calls bill incoming calls from PSTN to SIP using a special
22. provide certain law enforcement agencies with the ability to monitor calls of a certain subscriber 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 where you provide the service In PortaBilling you may activate the Legal Intercept call feature for every account which requires it obviously the control for this feature is only accessible from the administrator interface and not visible for the end user When it is done for every outgoing or incoming call to this account PortaSIP will be instructed to engage the RTP proxy regardless of the other NAT traversal settings and to produce a complete call recording of the conversation The call recordings then may be delivered to the law enforcement agency by any applicable means or you may even provide a real time access to the location on PortaSIP server where these files are stored Secure calling PortaSIP fully supports Secure Real time Transport Protocol SRTP according to RFC 3711 which provides confidentiality message authentication and replay protection to the voice traffic exchanged between IP phones VoiceVPN rating Voice virtual private network feature provides special handling of calls within a specific IP Centrex environment which typically represents telephony system for some enterprise Most of the features e g abbreviated dialing were previously discussed but ther
23. rate for routing at all e Huntstop signals that no routes with a lower preference should be considered This allows you to easily manage both termination costs and routing from a single location on the web interface Thus when such a routing tariff is associated with a connection you can send calls for termination to all prefixes for which rates exist in the tariff Multiple routes It is dangerous to have only one termination partner if it is down your customers will not be able make any more calls Normally you will try to find several vendors and enter their rates into the system Each connection to a vendor with routing tariff will produce one possible route and PortaBilling will arrange them according to cost or your other preferences 2000 2006 PortaOne Inc All rights reserved www portaone com 41 Porta SIP System Concepts Routing plans Routing preferences in the rate allow you to specify that for example you would rather send a call to MCI than to T Systems However this decision is global and so will apply to all calls made in your system But what if you would like to use MCI first for customer A while T Systems should be the first route for customer B and customer C should be routed to MCI only This can be accomplished using routing plans A routing plan defines the routes for which categories are available as well as in which order they should be arranged For instance in the example above
24. 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 2000 2006 PortaOne Inc All rights reserved www portaone com 40 Porta SIP System Concepts 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 Routing of SIP on net calls The SIP server automatically maintains information about all currently registered SIP user agents so it is able to determine whether a call should be sent directly to a SIP user agent Routing of off net calls You can have different vendors for terminating off net calls For example you can terminate calls to the US either to AT amp T via a T1 connected to your gateway in New York or to a remote gateway from Qwest Rate routing parameters Ordinarily tariffs define the termination costs for each connection to a vendor If you create a tariff with the Routing type a few more fields will be added to rates in that tariff e Route category you can split this into categories such as Premium Cheap etc and use these categories in routing plans e Preference routing priority 0 10 higher values mean higher priority 0 means do not use this
25. 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 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 diff
26. 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 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 concurrently 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 2000 2006 PortaOne Inc All rights reserved www portaone com 81 Porta SIP Administration FAQ 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
27. 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 SER 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 pairs separated by a colon s 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 LADDR SIP_PORT CANONIC_NAME PB_MASTER RAD_KEY Description IP address of the SIP environment Port on the SIP server which SIP phones should connect to value number default 5060 Pully 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 p
28. 006 PortaOne Inc All rights reserved www portaone com 92 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 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 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 2006 PortaOne Inc All rights reserved www portaone com 93 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 add
29. 03 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 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 2006 PortaOne Inc All rights reserved www portaone com 1 4 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 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 A s 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 per
30. 5060 portasipl SRV 10 0 5060 portasip2 SRV 60 0 5060 portasip3 2000 2006 PortaOne Inc All rights reserved www portaone com 27 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 attempt to contact them in sequence until it succeeds Advanced Features 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 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 c
31. All rights reserved www portaone com 43 Porta SIP System Concepts E a Test Dialplan gt Close Objects gt I Logout Phone Number Routing Plan protic CI Date and Time a H323 SIP YYYY MM DD HH Mi 861045676 Default v v search Destination Country Description Price Route Category Preference Huntstop Route to Vendor Connection Tariff 1 8610 CHINA Beijing 0 04000 USD Default 6 N ny gw 01 VendorA Termination to carrier A Vendor A 2 86 CHINA Proper 0 03000 USD Default 5 N ny gw 01 Vendor C Termination to carrier C Vendor C 3 86 CHINA Proper 0 06000 USD Default N ny gw 01 VendorB Termination to carrier B Vendor B 4 8610 CHINA Beijing 0 09000 USD Default 5 N 64672191 VendorF Termination to vendor F Vendor F 5 86 CHINA Proper 0 11000 USD Default N 193 50 123 6 VendorE Termination to carrier E Vendor E 6 86 CHINA Proper 0 02500 USD Default 2 N 45 12 156 200 Vendor D Termination to vendor D Vendor D Routing configuration example Tariff A Y gt Tariff B Tariff gt 8610 0 04 min 86 0 06 min 86 0 03 min Cheap 7 Default 5 Cheap 6 i N dl Q F Portals Billing te Termination Partner A Termination Termination Partner B A Partner C Tariff D 86 0 025 Cheap Consider the following example If you have 1 A Standard routing plan which includes thre
32. 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 3 A callis 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 In appendices A through C you will find a list of tested 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 single RTP proxy on an average PC server can support about 5 000 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 2000 2006 PortaOne Inc All rights reserved www portaone com 5 4 Porta SIP System Concepts SIP to SIP calls a Phone B d 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
33. MCI may be assigned as the Normal route category and T Systems as the Premium category After that three routing plans will be created e Quality includes first Premium and then Normal routing categories e Ordinary includes first Normal and then Premium routing categories e Cost efficient includes only Normal routing category So depending on which routing plan is assigned to the current customer the system will offer a different set of routes Routing algorithm The routing principle is simple e The SIP server or MVTS or some other entity asks PortaBilling for routing destinations for a given number e PortaBilling checks every tariff with routing extensions associated with a vendor connection for rates matching this phone number In each tariff the best matching rate is chosen this rate will define the routing parameters e A list of possible termination addresses will be produced this will include the remote IP addresses for VoIP connections and IP addresses of your own nodes with telephony connections e This list will be sorted according to routing plan routing preference and cost entries after the first huntstop will be ignored e If there are several routes with identical cost preference load sharing will be applied so that each potential route has an equal chance of being the first Consequently if your termination carrier provides you with three gateways to send calls to at the end of the day each
34. P 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 through 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 If the 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 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 PortaSIP Maintenance Release 12 introduced a new feature to follow me services simultaneous ringing Now 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 Follow me with the original DNIS CLD Very often a company operating an IP PBX would p
35. PORTA ONE Porta SIP i nd NS Administrator Guide Maintenance Release 13 www portaone com Porta SIP PortaSIP Administrator Guide Copyright notice amp disclaimers Copyright 2000 2006 PortaOne Inc All rights reserved PortaSIP Administrator Guide October 2006 Maintenance Release 13 V 1 13 1 Please address your comments and suggestions to Sales Department PortaOne Inc Suite 400 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
36. aSIP Administrator Guide APPENDIX A Tested Routers and NAT Software 87 APPENDIX B Cisco GW Setup for PortaSIP COMEDIA uu cscs 87 APPENDIX C Client s Cisco ATA 186 Configuration for PortaSIP 87 APPENDIX D Client s Sipura Configuration for PortaSIP oe 89 APPENDIX E Configuring Windows Messenger for Use as a SIP User FA a aerate tee ae ne re eae er See nr Re 91 APPENDIX F SJ Phone Configuration for PortaSIP 94 APPENDIX G Setting up a Back to Back T1 E1 Connection 96 APPENDIX H SIP Devices with Auto provisioning cccecsescesseseesnee 98 2000 2006 PortaOne Inc All rights reserved www portaone com 3 Porta SIP PortaSIP Administrator Guide Preface 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
37. 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 2000 2006 PortaOne Inc All rights reserved www portaone com 4 Porta SIP PortaSIP Administrator Guide Hardware and Software Requirements Server System Recommendations One UNIX Server A minimum of 40 GB of available disk space this space is required for storing various log files A processor running at 2 4 GHz or greater Additional processor speed is needed for networks with a high call volume Atleast 512MB of RAM RDRAM or DDR 1 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 1SO08859 1 books faq hardware html Client System Recommendations OS Windows 95 XP UNIX ot Mac OS Browser Internet Explorer 6 0 or higher Netscape 7 1 Mozilla 1 7 ot higher supporting DOM and with JavaScript enabled Spreadsheet processor MS Exce
38. 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 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 O Converts the phone number to the E 164 format so that 2027810003 become 12027810003 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 Starts the timer once the call is established disconnecting the call when the maximum call duration is exceeded 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 ad
39. 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 problematic 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 2006 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 As 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 e 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 connect
40. ash911 and their number will probably increase Naturally local E911 providers will be found in other countries as well To accommodate the demand for working with different providers PortaBilling uses a plugin model similar to that used for online 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 customers location Special handling of 911 calls Of course PortaBilling applies a special policy for processing and routing emergency calls For instance even if a cus
41. ate them to a telephony network 67 configure my Cisco gateway to send outgoing calls using SIP 68 configure my Cisco gateway for PSTN gt SIP service uo cece 69 Support incoming H323 and SIP calls on the same gateway 69 configure my Cisco ATA186 to work with PortaSIP 70 provide services to and bill a customer who has a SIP enabled gateway but no authorization capability e g Cisco AS5350 70 make all SIP calls to a certain prefix NNN go to my gateway XXX 70 allow my customer to have two phone numbers from different countries which will both ring on the same SIP phone 71 create an application to handle PSTN gt SIP calls 71 configure SIP phone X made by vendor Y 71 a BIlNSIP to S P calS irunia 72 bill incoming calls from PSTN to SIP using a special rate 72 provide error messages from the media server in my users local PAIN QUA Eininn aa iais 73 Calculate how much bandwidth need for my PortaSIP server 73 enable my SIP phone or ATA to be automatically provisioned by Porta SWEN isnie sonense nnna knn 74 3 Administration FAQ 75 Troubleshooting Common Problems 76 AO D MR 77 PortaSlP COMPGQUPALION ss hhininsdinananiehintomemmansan 82 4 APPENIC S Rails 86 2000 2006 PortaOne Inc All rights reserved www portaone com 9 Porta SIP Port
42. ave 2000 2006 PortaOne Inc All rights reserved www portaone com 23 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 Porta K Billing Billing Provisioning 1 Billing Engine SIP Registrations J PortaSIP PortaSIP In this case the call flow is exactly the same as in a situation where only one PortaSIP server is available discussed earlier in the SIP UA lt gt SIP UA chapter 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 2006 PortaOne Inc All
43. codec preference 10 g7llalaw codec preference 11 g7llulaw 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 2006 PortaOne Inc All rights reserved www portaone com 67 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
44. ctly 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 standard 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 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 k
45. dition Cisco gateways currently do not support INVITE with authorization 2000 2006 PortaOne Inc All rights reserved www portaone com 1 8 Porta SIP System Concepts e PortaSIP receives the INVITE but without authorization information So the PortaSIP 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
46. ds 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 or g729 codec since in this case you must pay a license fee A solution is to pre convert this voice prompt into a 8723 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 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 http www voip info org wiki 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 g720 codec you will need 31 2Kpbs 2 60 3 7Mbps Note that we 2000 2006 PortaOne Inc All rights reserved www portaone com 73 Porta SIP How to
47. e 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 2000 2006 PortaOne Inc All rights reserved www portaone com 33 Porta SIP System Concepts 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 the 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 e Cisco VoIP gateway any Cisco gateway running IOS this does not apply to Cisco ATA 186 188 e Nextone SBC e Sonus switch e Mera SIP HIT e 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 2000 2006 PortaOne Inc
48. e 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 E Billing Engine Billing A SIP Provisioning Registrations PortaSIP PortaSIP 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 2006 PortaOne Inc All rights reserved www portaone com 29 Porta SIP System Concepts Improved reliability Porta K Billing Billing Engine Billing SIP Provisioning Registrations PortaSIP PortaSIP PortaSIP Even if one of the SIP servers is down due to network issues or hardware problems your subscribers can continue using the service via other SIP servers Better network utilization You can install several SIP servers in different geographical locations as shown below with users within a certain network able 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 Qo Reke As EZ PortaBilling PortaSIP RE JA Master Sl
49. 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 then 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 2006 PortaOne Inc All rights reserved www portaone com 1 6 Porta SIP System Concepts Terminating SIP calls to a vendor using telephony Porta M Billing amp SIP phone A GW NY 02 Phone C 12 34 56 78 e Let s assume that T1 is connected to Qwest on our gateway GW NY 02 in New York where we ate 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 aut
50. 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 normal 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 2000 2006 PortaOne Inc All rights reserved www portaone com 99 Porta SIP System Concepts 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 e PortaSI
51. e charged for a call duration quite close to the real one First login greeting This is a feature not directly related to call processing but is something that will give your PortaSwitch based VoIP service a competitive 2000 2006 PortaOne Inc All rights reserved www portaone com 32 Porta SIP System Concepts 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 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 Digest authorization PortaSIP UA ser bZbua AAA server 70 68 0 213 216 231 44 34 216 231 44 34 time Sipura SPAZ000 3 1 5 Ss PortaSIP PortaBilling 8 Dec l 10 19 48 gt ta 101 1 INVITE gt 10 19 48 lt 100 tryin 10 19 48 G gt A 101 1 INVITE gt 10 19 48 lt a AE _ Unauthor l
52. e 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 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 2100 3 2 portasip2 IN A 193 100 375 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
53. e is one important issue remaining how these calls will be charged We need to have a consistent way to charge all calls between IP phones of the customer regardless of the actual phone number dialed for instance a customer may have phone numbers from the different countries If a Voice VPN feature is enabled for a particular customer when the call is made from an account A which belongs to this customer to an account B which belongs to the same customer PortaBilling will look up the applicable rate not for the actual phone number but for the special keyword voiceven and use it for charging the call So by placing into the tariff applied to your customers a rate to that destination you may specify 2000 2006 PortaOne Inc All rights reserved www portaone com 39 Porta SIP System Concepts how such calls should be rated should they be free or charged some nominal amount etc Using the vorcEven rate in tariffs allows you to avoid having sip to sip minutes mixed in with the off net minutes in case Associated feature is Voice VPN Distinctive Ring If acticated when the call arrives from one of the IP phones within the same IP Centrex environment PortaSIP will send an instruction to the IP phone to use a ringing pattern different from the default If the phone supports distinctive ringing this allows the end user to immediately recognize if the call is coming from one of his colleagues or from some exte
54. e route categories Default order 70 Cheap order 40 and Expensive order 10 2 Six vendors A B C D E F with the following rates prefix route category preference price a 8610 Cheap 7 0 04 b 86 Default 5 0 06 c 86 Cheap 6 0 03 d 86 Cheap 6 0 025 e 86 Expensive 5 0 11 f 8610 Premium 5 0 09 then when a customer with this routing plan makes a call to 8610234567 the system will arrange the possible routes as follows Vendor Parameters Comment B Default 5 0 06 The Default route category is first in the route plan A Cheap 7 0 04 This vendor has the highest preference in the Cheap category D Cheap 6 0 025 This vendor has the same preference as vendor C but a cheaper per minute rate 2000 2006 PortaOne Inc All rights reserved www portaone com 44 Porta SIP System Concepts C Cheap 6 0 03 E Expensive 5 0 11 This is the only vendor in the last route category Vendor F was not included in the routing since his route category is not in the customer s routing plan ea 2 Test Dialplan gt Close Objects PT Logout Protocol Date and Time Phone Humber Routing Plan _ H323 SIP YYYY MM DD HH Mi 8610234567 Standara v v Destination Country Description Price Route Category Preference Huntstop Route to Vendor Connection Tariff 1 86 CHINA Proper 0 06000
55. e 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 2006 PortaOne Inc All rights reserved www portaone com Porta SIP technology inc j T System SIP Provisioning SA Line 2000 2006 PortaOne Inc All rights reserved www portaone com 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 SRV 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 0xb8 5060 none v no x yes M 216 231
56. e way conference call Supported by PortaSwitch SIP phone have to 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 2006 PortaOne Inc All rights reserved www portaone com 65 Porta SIP 2 s How to 2000 2006 PortaOne Inc All rights reserved www portaone com How to 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
57. eep 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 2000 2006 PortaOne Inc All rights reserved www portaone com 51 Porta SIP System Concepts endpoint clients fill in this information 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 in ae aa 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
58. erent 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 cetrex user can upload his own melody or use the default one for his centrex environment Speed Dialing Feature description Allows the user to call frequently called telephone numbers by dialing an abbreviated speed calling code instead of the entire number 2000 2006 PortaOne Inc All rights reserved www portaone com 6 4 Porta SIP System Concepts Supported by PortaSwitch via the Abbrevited 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 thre
59. f 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 on a private 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 IP 9 8 7 6 Port 12345 IP 192 168 0 7 Port 56789 H A 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 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
60. ferences same preference for all vendors then the routing decision will be affected solely by the vendor s cost Compare the two illustrations below The first one shows simple least cost routing all of the available carriers are arranged according to cost In the second illustration PortaSwitch routing preferences are used so that the administrator can re arrange the routing In this case carrier D has been moved down the routing list while carrier A has been moved up to the top of the list thus becoming the first route WH Test Dialplan gt Close 4E Objects gt I Logout K Protocol Date and Time Phone Number Routing Plan H323 SIP YYYY MM DD HH Mi 8610234567 Default v v Search Destination Country Description Price Route Category Preference Huntstop Route to Vendor Connection Tariff 86 CHINA Proper 0 02500 USD Default 5 45 12 156 200 Vendor D Termination to vendor D Vendor D 86 CHINA Proper 0 03000 USD Default 8610 CHINA Beijing 0 04000 USD Default 86 CHINA Proper 0 06000 USD Default 8610 CHINA Beijing 0 09000 USD Default 86 CHINA Proper 0 11000 USD Default N N ny gw 01 VendorC Termination to carrier Vendor C N ny gw O1 VendorA Termination to carrier A Vendor A N ny gw 01 VendorB Termination to carrier B VendorB N 64 67 2191 VendorF TerminationtovendorF VendorF N 193 50 123 6 VendorE Terminationto carrierE VendorE Dne wn aana 2000 2006 PortaOne Inc
61. forms 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 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 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 2006 PortaOne Inc All rights reserved www portaone com 1 5 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
62. h both incoming SIP and H323 calls even if it gives the session protocol sipv2 dial peer voice 101 voip description Incoming SIP calls incoming called number 2000 2006 PortaOne Inc All rights reserved www portaone com 69 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 APPENDIX C Client s Cisco ATA 186 Configuration for PortaSIP 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 fo
63. hentication 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 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 2006 PortaOne Inc All rights reserved www portaone com 1 7 Porta SIP System Concepts PSTN gt SIP Phone C ig NY 01 2 Porta Porta E4Billing Porta E4Billing SIP ga 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
64. here is no answer after a specified number of rings This feature is implemented by provisioning the Follow me service 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 Feature description Allows user to place call on hold move to a different location and then resume the call from any other station in the centrex by dialing a pickup code Supported by PortaSwitch 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 is available covering incoming
65. horization 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 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 tecognizes that the call should be sent to one of your SIP phones 7 Follow me UM parameters and other related 2000 2006 PortaOne Inc All rights reserved www portaone com 3 8 Porta SIP System Concepts 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
66. hts reserved www portaone com 25 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 E Eo Porta K Billing Billing Provisioning Billing Engine 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 Engine Es ng CES Eo ng E EN e LED T PortaSIP NM 7 e FD Re Ke ZS LILES 2000 2006 PortaOne Inc All rights reserved www portaone com 26 Porta SIP System Concepts SIP phone configuration for PortaSIP cluster In order to ensure reliable VoIP services 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 hostnam
67. iginally dialed number DNIS When entering the follow me information the destination should be entered with prefix e g 12061234567 This will instruct PortaSwitch that the call must be forwarded to 12061234567 destination here to the registered SIP phone with this number but in the INVITE message the To should contain the original DID Then the IP PBX will properly process the incoming calls and forward them to the correct recipient Service announcements via the media server customer might be unable to make 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 that 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 Ho
68. in the system as a trusted node configure my 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 2006 PortaOne Inc All rights reserved www portaone com 68 Porta SIP How to session target sip server l You probably will need an application on the incoming telephony
69. ion 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 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 2006 PortaOne Inc All rights reserved www portaone com 20 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 Please contact the PortaOne support team for assistance with this A configuration task since if you configure the network interface on the SIP server improperly it will render all of your SIP services useless _
70. it is possible to apply this rule to convert the number When billing returns a list of routes the phone numbers for routes for this connection will be converted This only works for a routing model in which the VoIP node e g PortaSIP requests billing for routing information If your gateway uses dial peers or an external gatekeeper for routing then you must configure number translation there Connection based translation rules When the call has been terminated to the vendor in a vendor specific format it will be reported to billing in this same format e g 7834 42021234567 Now it is necessary to convert this number to the proper format for billing 4202134567 which may be done using connection translation rules These rules will be applied to all calls which go through a given connection even those routed there using dial peers ot other external tools Node based translation rules These serve the purpose of converting a number from a custom format used by the customer into billing s internal format during authorization depending on the gateway For example on a gateway in Prague Czech Republic there may be the translation rule strip leading 00 while on a gateway in Moscow Russia the rule will be strip leading 810 or replace leading 8 with 7 Since customer based translation rules were introduced node based translation has become obsolete Therefore a node based translation rule is applied only if there is n
71. l 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 2006 PortaOne Inc All rights reserved www portaone com 5 Porta SIP PortaSIP Administrator Guide 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 1 stable branch with the latest security bug fixes supplementary packages necessary for convenient system administration and maintenance 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 13 The Release includes several new features and improvements e Support for SIP TAPI applications e Support for music on hold using g723 codec e Improved performance of RTP proxy e Increased maximum number of simultaneously registered SIP phones per PortaSIP server 50 000 IP phones per server
72. ler ID Caller ID on Call 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 Intercom Dialing Feature description Allows users to call Centrex extensions by dialing a standard 4 digit code instead of the entire 7 digit telephone number Supported by PortaSwitch via the Abbreviated Dialing feature 2000 2006 PortaOne Inc All rights reserved www portaone com 63 Porta SIP System Concepts 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 Follow me feature Message Waiting Audible Feature description Provides
73. lity 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 do not want to use an RTP proxy since it will increase the amount of required bandwidth can use STUN instead The STUN REC http www faqs org rfcs rfc3489 html states This protocol is not a cure all for the problems associated with NAT 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
74. local erc 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 2006 PortaOne Inc All rights reserved www portaone com 85 Porta SIP 4 Appendices 2000 2006 PortaOne Inc All rights reserved www portaone com Appendices Porta SIP Appendices APPENDIX A Tested Routers and NAT Software Commodity routers and NAT software bundled with popular operating systems which attempt to preserve the RTP source port Linksys BEFSX41 Belkin F5D5230 4 natd bundled with FreeBSD 4 x and 5 x operating systems iptables bundled with Linux kernel 2 4 x Tp o Commodity routers and NAT software bundled with popular operating systems which do not attempt to preserve the RTP source port 1 Internet connection sharing software bundled with the Windows XP operating system 2 Netgear RP614 APPENDIX B Cisco GW Setup for PortaSIP COMEDIA sip ua nat symmetric check media src APPENDIX C Client s Cisco ATA 186 Configuration for PortaSIP UIDO CLIENT S ACCOUNT ID PHONE NUMBER 1 PWDO CLIENTS PASSWORD FOR ACCOUNT ID 1 UIDI CLIENT S ACCOUNT ID PHONE NUMBER 2 PHDI CLIENTS PASSWORD FOR ACCOUNT ID 2 GkOxProxy IP ADDRESS OF SERVER RUNNING PORTASIP Gateway 0 0 0 0 GateWay2 0 0 0 0 seLoginID 0 LoginIDO 0 LoginID1 0 AltGK 0 0
75. nation Unattended blind transfer Porta 4 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 2000 2006 PortaOne Inc All rights reserved www portaone com 35 Porta SIP System Concepts 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 Under account A A gt B 15 minutes e Under account B A gt C 10 minutes As a result A doe
76. nd C together and a REFER message is sent to PortaSIP 10 e PortaSIP will now connect A and C together 12 and cancel both of the call legs going to B e 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 2000 2006 PortaOne Inc All rights reserved www portaone com 37 Porta SIP System Concepts 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 g A S asm DID Provider Eo 2 _ aT Is port sie TT porta Lemno 7 Re 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 aut
77. ne 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 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 username 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 2000 2006 PortaOne Inc All rights reserved www portaone com 80 Porta SIP Administration FAQ 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 functiona
78. nsure 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 vendor and set up the NAT traversal status accordingly d na NAT traversal oe a Porta KE SIP 2 Vendor A I as L e Pre go es NAT traversal NAT_ gt Ss avajlable call 2 RTP VendorB NX 7 s a _ 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 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 Error Reference source not found for details on RTP proxy policy configuration 2000 2006 PortaOne Inc All rights reserved www portaone com 56 Porta SIP System Concepts Auto provisioning of IP phones If
79. o customer based translation rule defined for a given customer 2000 2006 PortaOne Inc All rights reserved www portaone com 46 Porta SIP System Concepts Number translation on your network Below is an illustration of how different translation rules are applied during a call Porta Billing 100 customer rate amp routing connection dialing rule lookup outgoing rule 0042021234567 gt 42021234567 gt 01142021234567 ROUTING AIRES MONI IP address 1 2 3 4 003208123350 01142021234567 Porta SIP 0042021234567 01142021234567 6 Number inside of your VolP is represented as RZ 42021234567 Customer s IP Phone Cell Phone Carrier ABC 1 The customer dials a phone number on his SIP phone He enters the number in the same format he uses on his conventional phone i e 0042021234567 2 The number is delivered to the PortaSIP server and translated using the customer s dialing rule which states that the international dialing prefix for this customer is 00 So the number becomes 42021234567 E 164 format This number is used to search for the customet s rate for this destination 3 PortaSIP then requests routing for this number Carrier ABC is defined for terminating calls to the Czech Republic in PortaBilling However this carrier requires the number to be in US dialing format so the international number must be prefixed by 011 An outgoing translation rule s 011 to carrier ABC
80. offline due to a network outage e Various IP Centrex features call waiting call hold music on hold abbreviated dialing follow me etc e Fail over routing a list of routes arranged according to cost preference and customer routing plan is supplied by PortaBilling100 e Forwards calls to the unified messaging service PortaUM if a SIP phone is not available 2000 2006 PortaOne Inc All rights reserved www portaone com 1 0 Porta SIP System Concepts PortaSIP Components Porta K Billing100 2 Provisioning Server sr o RADIUS Client Taunt SIP registrar proxy eons ai RADIUS Client e LS RADIUS w RTP SIP Porta Proprietary PortaSIP components e SIP Proxy Server SIP Express Router on the diagram 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
81. one itself The device will periodically go to the provisioning server and fetch its configuration file Cisco ATA Expert This utility allows you to simplify manual provisioning of a Cisco ATA 186 188 device and browse the device configuration in a user friendly format For example instead of entering values such as 0 1 2 3 etc for codec selection you can choose names such as G 729 from a select menu This tool is convenient for single time configuration of a device or for troubleshooting 2000 2006 PortaOne Inc All rights reserved www portaone com 57 Porta SIP System Concepts 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 configura
82. onnection 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 2006 PortaOne Inc All rights reserved www portaone com 9 8 Porta SIP System Concepts Follow me services Due to the volatile nature of VoIP networks the customer may wish to use standard PSTN calls as a backup He can define a list of follow me numbers for each of which a period of validity can be defined e g he wants to receive calls to his mobile phone only from 8am to 9pm When a call arrives on his original SIP account the SIP server can try the alternative numbers until the call is answered Porta 4 Billing GW NY 01 Phone C SIP es A 2 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
83. orta admin pl utility on the slave server to find the correct value RADIUS configuration IP address of the PortaBilling100 master host RADIUS secret key for RADIUS requests to the billing value strin 2000 2006 PortaOne Inc All rights reserved www portaone com 82 Porta SIP Administration FAQ Variable Description AUTH_PORT Port on the RADIUS server to which authentication requests should be sent 1812 by default ACCT_PORT Port on the RADIUS server to which accounting RAD_TIMEOUT RAD_RETRIES 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 FIRSTLOGIN_CLI B2B_KA A B2B_KA O SEND_START_ACCT MAX_CREDIT_TIME HUNT_STOP REG_EXPIRES MIN REG_EXPIRES MAX Variable Description FIRSTLOGIN_ENABLE Activate the first login greeting feature 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 Send keep alive requests to the caller party originating SIP device possible values 0 ot 1 Send keep alive requests to the called party terminating SIP device possible values 0 or 1 Send an accounting request to the billing when the call is started this is necessary if y
84. ortaBilling 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 Keep alive call monitoring If 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 Thus 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 available e Customer 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 b
85. ou 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 2000 2006 PortaOne Inc All rights reserved www portaone com 83 Porta SIP Administration FAQ ALLOW_ASYMMETRIC 0 or 1 1 forces an RTP asymmetric flag for any non NAT UA The default is 0 PROCESS REFER 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 2006 PortaOne Inc All rights reserved www portaone com 8 4 Porta SIP Administration FAQ Starting Stopping PortaSIP Services If you need to stop all PortaSIP services then execute the following command sudo usr local erc rc d sip sh stop This will properly terminate all components To start PortaSIP use the following command sudo usr
86. ount 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 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 2006 PortaOne Inc All rights reserved www portaone com 1 9 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 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 use
87. our customer will have one main account e g 12027810003 which will be provisioned 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 from http store portaone com 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 2006 PortaOne Inc All rights reserved www portaone com 71 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
88. outers 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 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 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
89. r i e 2027810003 This is accomplished via the CLI translation rule property of the vendor s connection CLI translation rules calls terminated to SIP phones Another extremely useful feature of the CLI translation rule is PortaSwitch s ability to convert the CLI ANI number for the incoming call into the customer s dialing format activated in the customer s dialing rules settings Let s assume that a customer has a SIP phone with the phone number 12027810003 provisioned to it and his dialing rules are setup for North America While out for lunch he receives three calls e From phone number 12027810002 his colleague e From 420298765432 his customer in the Czech Republic e From 12061234567 his old friend from Seattle The ANI CLI numbers for all these calls will be converted so that when he returns from lunch he will see e 7810002 e 011420298765432 e 12061234567 Now he can simply hit redial on his phone to initiate a call since these numbers are already in the same format as he would have normally dialed Routing of SIP On net Calls The SIP server automatically maintains information about all currently registered SIP user agents Thus it is able to determine how to contact a specific SIP user agent if there is an incoming call In response to the authorization request the billing engine informs the SIP server that the dialed number is actually a valid SIP account and that the call should be
90. r 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 actually 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 1 s server is down or for some reason inaccessible 2000 2006 PortaOne Inc All rights reserved www portaone com 1 3 Porta SIP System Concepts Porta M Billing SIP phone A SIP phone B In the example above user A assigned SIP phone number 120278100
91. r user agents i e for devices which represent a single physical phone An SIP 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 2000 2006 PortaOne Inc All rights reserved www portaone com 70 Porta SIP How to e When you enter rates into this tariff two new columns will appear Preference and Huntstop Enter the desired routing 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 Y
92. rate 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 2000 2006 PortaOne Inc All rights reserved www portaone com 12 Porta SIP How to 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 provide error messages from the media server in my users local language First 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 sln Upload the files to the PortaSIP server in the usr local share asterisk soun
93. re 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 P5 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 2000 2006 PortaOne Inc All rights reserved www portaone com 61 Porta SIP System Concepts This feature is implemented by provisioning the Follow me service 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 t
94. ress of the person 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 om APPENDIX F SJ Phone 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 1 Service PortaOne Please enter this information to initialize the service profile Lax oo canca Password 0 Save service information permanently 2000 2006 PortaOne Inc All rights reserved www portaone com 9 4 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 2006 PortaOne Inc All rights reserved www portaone com Porta SIP Appendices APPENDIX G Setting up a Back to Back T1 E1 Connection Hardware Setup In order to make one or more back to back connections you will need to construct one or more RJ 48C cross over cables using the following table T1 E1 CSU DSU Cross Over Pinout From RJ 48C Pin To RJ 48C Pin 1
95. ress 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 phones 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 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
96. rights reserved www portaone com 13 Porta SIP Administration FAQ APPENDIX G Setting up a Back to Back T1 E1 Connection you will find information on how to set up such a back to back connection physically and configure it in Cisco IOS 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 be 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 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 Cisco ATA 186 for other SIP phones or gateways check the documentation supplied with the device If you ate 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 determi
97. rights reserved www portaone com 9 4 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 rorta K Billing Billing Engine n E Eo Sn 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 this 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 2006 PortaOne Inc All rig
98. rnal number IP Centrex feature management The question of convenient and efficient service provisioning becomes vety important when managing an IP Centrex hosted IP PBX environment with tens or even hundreds of IP phones in it In case if you need to change some parameter e g CLI number for outgoing calls for all of the IP phones it is highly desireable to avoid the situation when you have to change this parameter manually on every account PortaSwitch splits call feature management in two parts e Some parameters are defined on the customer level so they are global for the whole IP Centrex environment of this customer e You may also manage the call features on the account level You have an option to either manually override a certain parameter s value or to specify that the current value defined on the customer level should be used This allows you to define most of the call features parameters only once on the customer level and they will be automatically propagated to the level of accounts individual phones 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 to choose the best route least cost routing When a call goes through the PortaSIP server
99. s 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 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 4 Billing 5 PSTN GW Phone C 2 8 gat gis 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 2000 2006 PortaOne Inc All rights reserved www portaone com 36 Porta SIP System Concepts Attended transfer Porta Billing SIP phone A SIP phone B 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 send a call to this destination and to obtain the routing 7 In the case of a positive reply PortaSIP establishes a call to C 8 e The call is now established between B and C 9 after a short exchange B decides to bridge A a
100. sent to the SIP user agent Note that routing the call to a SIP user agent is only one of the possible routes for instance a call can be redirected to 2000 2006 PortaOne Inc All rights reserved www portaone com 48 Porta SIP System Concepts follow me numbers or a unified messaging service if the account is not available online at the moment Routing of SIP Off net Calls You can have different vendors for terminating off net calls For example calls to the US can be terminated either to AT amp T via a T1 connected to your gateway in New York or by sending the call to a remote gateway from Qwest You need a tool allowing you to manage routing policies for the different destinations This tool is extensions routing for tariffs Tariffs define the termination costs for each connection to a vendor while extensions routing simply adds a few more fields to the rates in a given tariff This allows you to easily manage both termination costs and routing from a single location on the web interface The routing principle is simple e The SIP server asks PortaBilling for routing destinations for a number e PortaBilling checks every tariff with routing extensions associated with connection to the vendor for rates matching this phone number e A list of possible termination addresses will be produced this will include remote IP addresses for the VoIP connections and IP addresses of your own nodes with telephony connections e This lis
101. 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 possible 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 2000 2006 PortaOne Inc All rights reserved www portaone com 50 Porta SIP System Concepts 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 are 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 dire
102. source ports from which SIP messages arrive and correct these as well This allows SIP signaling to 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 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 2000 2006 PortaOne Inc All rights reserved www portaone com 59 Porta SIP System Concepts 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 r
103. ss isdn switch type primary 5ess isdn protocol emulate network no cdp enable interface Serial3 23 no ip address isdn switch type primary 5ess no cdp enable 2000 2006 PortaOne Inc All rights reserved www portaone com 97 Porta SIP Appendices APPENDIX H 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 WRT54GP2 e GrandStream HT486 e GrandStream HT496 2000 2006 PortaOne Inc All rights reserved www portaone com 98
104. t will be sorted according to the routing preference with entries after the first huntstop being ignored e A list of these IP addresses with optional login and password for SIP authentication will be returned to the SIP server NAT Traversal Guidelines NAT Overview The purpose of NAT Network Address Translation is to allow multiple hosts on a private LAN not directly 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 2000 2006 PortaOne Inc All rights reserved www portaone com 49 Porta SIP System Concepts 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 o
105. 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 2006 PortaOne Inc All rights reserved www portaone com 76 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 hardware 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
106. tion files will be built for all user agents Each user agent will then retrieve this file the next time it goes online Porta aK Billing Provisioning server Account phone line T Phone Passed p prone confi iP phone inventory record i MAC address gt aa 1 rs l IP phone profile General parameters B L y ie Request for provisioning information gt g Configuration fle gt 39 000 OOO 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 2000 2006 PortaOne Inc All rights reserved www portaone com 5 8 Porta SIP System Concepts 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 add
107. 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 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 2006 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 K 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 acc
108. tomer s account has exceeded its balance and he cannot make outgoing calls a 911 call will still go through 2000 2006 PortaOne Inc All rights reserved www portaone com 60 Porta SIP System Concepts Interconnection with an E911 provider Two steps are involved here e Connecting to the E911 provider s API to validate and populate 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 IP Centrex features This section provides a general overview of various IP 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 Anonymous Call Rejection Featu
109. urchase multiple phone numbers which should all be routed to the company e g the main office phone number is in the New York area but the company also has a 1800 number and a number in the UK for their UK based sales representative In general each additional phone number will be provisioned as an account in PortaBilling and then a corresponding SIP phone will register to PortaSwitch using this account ID to receive the 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 through the follow me feature For example IP PBX registers to PortaSwitch with account 12061234567 but DIDs 18007778881 and 4412345678 must also be delivered to the IP PBX You would set up accounts 18007778881 and 4412345678 with the follow me to 12061234567 Then all calls will be correctly routed to the IP PBX but 2000 2006 PortaOne Inc All rights reserved www portaone com 30 Porta SIP System Concepts since they all arrive to the IP PBX as calls to 12061234567 calls to different DIDs cannot be distinguished e g when a customer originally dialed 1800 number he should be connected to the general sales and if a UK number was originally dialed this call should be answered by a specific group in the sales team In this situation a special follow me mode is required to preserve the or
110. w initial and operating costs that cannot be matched by yesterday s circuit switched and narrowband service provider PSTN 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 2006 PortaOne Inc All rights reserved www portaone com 9 Porta SIP System Concepts PortaSIP functions J L Termination gt Termination C partner A partner B Les Es i Bank Online payment processor Porta Billing M Administrator interface Residential IP Admin Web Q Self care _ Pre paid cards D ANVDNIS Termination Callback to PSTN Customized VR Porta UM Unified Messaging Phone amp Web Interface El ao 200 ooo 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
111. wever 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 e account_expired the account is no longer active expired as per the expiration date or life time 2000 2006 PortaOne Inc All rights reserved www portaone com 31 Porta SIP System Concepts e cld_blocked there was an attempt to call a destination which is not in the tariff or is marked as forbidden e ctedit_disconnect a call is disconnected because the maximum credit time is over e 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 uset_ denied the account is blocked e wrong passwd an incorrect password has been provided Every account in P
112. 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 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 2000 2006 PortaOne Inc All rights reserved www portaone com 59 Porta SIP System Concepts 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 D
113. 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 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 the 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 ph

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