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<p class="MsoNormal">Hi Joel,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have tried this along with setting outbound_proxy. Client2 sees the call from Client1, I click on “Answer” and the call eventually just drops, although the camera showing myself does come on after I click on “Answer”. Others on this
board have given it a real try to help, but I fear I am missing some deep concept. Note: I have looked at tcpdumps and asterisk logs for a few weeks now!
<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Just for grins, here are my test cases. <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">For the sake of making things a little easier to read I will define –<o:p></o:p></p>
<p class="MsoNormal">Asterisk Server: AS<o:p></o:p></p>
<p class="MsoNormal">Kamailio Server KS<o:p></o:p></p>
<p class="MsoNormal">WebRTC Client1 and Client2 are WebRTC (tryit) instances both using chrome<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>Test1</b> (tryit configured to go to AS via wss): Client1 can call Client2; Client2 answers and I can see Audio/Video on both Client1 and Client2 (<b>Success</b>!);<o:p></o:p></p>
<p class="MsoNormal">The tcpdump (AS) shows <b>no</b> SIP, RTP, SRTP activity, but a lot of UDP activity.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>Test2</b> (tryit configured to go KS via wss): Client1 calls Client2, Client2 rings and show’s “ANSWER” and after I answer Client1 hangs up (<b>Failure</b>!)
<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><b>Test3</b> (tryit configured to go KS via wss): Client1 calls an Asterisk Dial-Plan which plays audio and video (<b>Success</b>!)<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><a name="_MailEndCompose"><o:p> </o:p></a></p>
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<p class="MsoNormal"><b>From:</b> sr-users [mailto:sr-users-bounces@lists.kamailio.org]
<b>On Behalf Of </b>Joel Serrano<br>
<b>Sent:</b> Friday, December 15, 2017 2:22 PM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org><br>
<b>Subject:</b> Re: [SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Hi Steve, <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Sorry, I was replying at the same time in another thread and I mixed responses.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Let me correct myself: <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">You are using:<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-size:9.5pt"> same => n,Dial(PJSIP/${EXTEN},30,t)</span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">So if someone dials "client1" you would try to reach PJSIP/client1 and that would fail because client1 is not registered with Asterisk.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Let's say your trunk with Kamailio si called "kamailio" in pjsip.conf...<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">If "client1" is registered with Kamailio instead of Asterisk, you want to send that call to Kamailio:<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-size:9.5pt;font-family:"Courier New""> same => n,Dial(PJSIP/${EXTEN}<span style="color:red">@kamailio-trunk</span>,30,t)</span><o:p></o:p></p>
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<p class="MsoNormal"><span style="font-size:9.5pt;font-family:"Courier New""> same => n,Dial(PJSIP/<span style="color:red">kamailio-trunk/</span>${EXTEN},30,t)</span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span style="font-size:9.5pt">That way, you are sending the call to Kamailio instead of trying to keep it local.</span><o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Sorry for the confusion, let me know if you have any doubts.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Joel.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Fri, Dec 15, 2017 at 10:43 AM, Joel Serrano <<a href="mailto:joel@gogii.net" target="_blank">joel@gogii.net</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal">Hi Steve, <o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">You mentioned that Kamailio was handling the registration of the clients...<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Therefor, you want to do something like this if you want to be able to bridge calls:<o:p></o:p></p>
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<p class="MsoNormal"><br>
exten => _900XX,1,Answer()<br>
same => n,DumpChan()<br>
same => n,Dial(SIP/${EXTEN}@<<kamailio-peer>>,30,t)<br>
same => n,HangUp()<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Another valid format:<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">exten => _900XX,1,Answer()<br>
same => n,DumpChan()<br>
same => n,Dial(SIP/<<kamailio-peer>>/${EXTEN},30,t)<br>
same => n,HangUp()<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Where "<<kamailio-peer>>" == the name of the peer you configured in sip.conf for Kamailio.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">At the end, the key here is that you are calling SIP/XXXX which is not local, if you call PJSIP/XXXX (per your example) you are trying to reach a local endpoint (in this case, the device that would have to be directly registered with Asterisk).<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Give a try and let me know how it goes.<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">Cheers, <o:p></o:p></p>
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<p class="MsoNormal"><span style="color:#888888">Joel.<o:p></o:p></span></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Thu, Dec 14, 2017 at 8:59 AM, Wilkins, Steve <<a href="mailto:swwilkins@mitre.org" target="_blank">swwilkins@mitre.org</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Hello,<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Here is my extentions.conf<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">exten => _900XX,1,Answer()<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,DumpChan()<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,Dial(PJSIP/${EXTEN},30,t)<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> same => n,HangUp()<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I have 90001,90002 in pjsip.conf with a webrtc endpoint.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thank you,<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">-Steve<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><a name="m_7261412455163425338_m_-767349331587823"> </a><o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"><b>From:</b> sr-users [mailto:<a href="mailto:sr-users-bounces@lists.kamailio.org" target="_blank">sr-users-bounces@lists.kamailio.org</a>]
<b>On Behalf Of </b>Joel Serrano<br>
<b>Sent:</b> Thursday, December 14, 2017 9:29 AM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Hi, can you share with us the asterisk dialplan part where you call the Dial() application?<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">On Tue, Dec 12, 2017 at 06:38 Wilkins, Steve <<a href="mailto:swwilkins@mitre.org" target="_blank">swwilkins@mitre.org</a>> wrote:<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Hello All,<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I am looking for a Diagram or such that shows the flow of SIP traffic for a WebRTC Client1 => WebRTC Client2 call using Kamailio in front of Asterisk.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">I am unable to get Asterisk to find the correct registered clients, which are registered in Kamailio and am hoping verifying the flow will help give me a clue as to what is going
on. E.g. Using chrome and tryit-pjsip I have Client1, and Client2 registered in Kamailio. However when I try to connect Client1 to Client2 (make a call), Asterisk has no clue where Client1 and Cleint2 are registered to.<o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto"> <o:p></o:p></p>
<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">Thank you!<o:p></o:p></p>
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<p class="MsoNormal" style="mso-margin-top-alt:auto;mso-margin-bottom-alt:auto">_______________________________________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></p>
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_______________________________________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></p>
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