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<p class="MsoNormal"><span style="mso-fareast-language:EN-US">Hello,<o:p></o:p></span></p>
<p class="MsoNormal"><span style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">one cause for this 30s call ending could be that the ACK for the 200OK on the INVITE is not routed correctly. Have a look to the signalisation of the clients and/or server.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">One way audio can be caused from wrong SDP, check that the media IPs in the SDB are correct.<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Cheers,<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Henning<o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">-- <o:p>
</o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Henning Westerholt –
</span><span style="mso-fareast-language:EN-US"><a href="https://skalatan.de/blog/"><span lang="EN-GB" style="color:#0563C1">https://skalatan.de/blog/</span></a></span><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p></o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US">Kamailio services –
</span><span style="mso-fareast-language:EN-US"><a href="https://gilawa.com/"><span lang="EN-GB" style="color:#0563C1">https://gilawa.com</span></a></span><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p></o:p></span></p>
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<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
<p class="MsoNormal"><span lang="EN-GB" style="mso-fareast-language:EN-US"><o:p> </o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><b><span lang="EN-GB">From:</span></b><span lang="EN-GB"> sr-users <sr-users-bounces@lists.kamailio.org>
<b>On Behalf Of </b>Kelvin Ezenwaka<br>
<b>Sent:</b> Friday, May 13, 2022 12:46 PM<br>
<b>To:</b> users <users@lists.kamailio.org>; sr-users-owner <sr-users-owner@lists.kamailio.org><br>
<b>Cc:</b> ope@tosoplimited.com; Ope Shokunbi <ope@pressone.co><br>
<b>Subject:</b> [SR-Users] Failing Audio stream on Kamailio+webrtc with integration to asterisk server<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span lang="EN-GB"><o:p> </o:p></span></p>
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<span style="font-size:10.5pt;font-family:"Segoe UI",sans-serif;color:#24292F">Summary:<o:p></o:p></span></p>
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<span style="font-size:10.5pt;font-family:"Segoe UI",sans-serif;color:#24292F">We have a Kamailio server as our sip proxy server, sip firewall with WebSocket and RTP engine configured on it (the Kamailio Server). But we experience one way audio, no audio, hang
up after 30s, when we try making calls between internal extensions (eg extension 100 to call extension 105) and external calls (eg extension 100 to call mobile number 09056925668) as well. The call flow is as:<o:p></o:p></span></p>
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<span style="font-size:10.5pt;font-family:"Segoe UI",sans-serif;color:#24292F">Webrtc client<------->Kamailio1+rtpengine<-----Asterisk---->Kamailio2<-------->Telco Provider<o:p></o:p></span></p>
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<span style="font-size:10.5pt;font-family:"Segoe UI",sans-serif;color:#24292F">The asterisk communication between all three boxes is via local IP (All ports open between them). The Kamailio 1 box where we have our sip registration cache, has rtp ports and wss
port open on the internet.<o:p></o:p></span></p>
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<span style="font-size:10.0pt;font-family:"Verdana",sans-serif">Asterisk 18.9.0<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:10.0pt;font-family:"Verdana",sans-serif">Kamailio 5.5.3 + RTPengine 10.4.0.0<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><span style="font-size:10.0pt;font-family:"Verdana",sans-serif">Debian 11 bullseye<o:p></o:p></span></p>
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<span style="font-size:10.5pt;font-family:"Segoe UI",sans-serif;color:#24292F">Kindly see attached below a diagram depicting the VoIP network flow and also attached are logs files for the webrtc client and server side.<o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt"><b><span style="font-size:10.0pt;font-family:"Helvetica Neue",serif;color:#1D2228;background:white">Ezenwaka Kelvin Arinze</span></b><span style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p></o:p></span></p>
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<p class="MsoNormal" style="margin-left:35.4pt;background:white"><span style="font-size:10.0pt;font-family:"Helvetica Neue",serif;color:#1D2228">VOIP Engineer<o:p></o:p></span></p>
<p class="MsoNormal" style="margin-left:35.4pt;background:white"><span style="font-size:10.0pt;font-family:"Helvetica Neue",serif;color:#1D2228">Tel: (+234) 9056925668<o:p></o:p></span></p>
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