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List: sr-users
Subject: Re: [SR-Users] Failing Audio stream on Kamailio+webrtc with integration to asterisk server
From: Henning Westerholt <hw () gilawa ! com>
Date: 2022-05-16 7:57:55
Message-ID: AM6PR05MB54093BE101C99EA35C14A1F2BFCF9 () AM6PR05MB5409 ! eurprd05 ! prod ! outlook ! com
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Hello,
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.
One way audio can be caused from wrong SDP, check that the media IPs in the SDB are \
correct.
Cheers,
Henning
--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>
From: sr-users <sr-users-bounces@lists.kamailio.org> On Behalf Of Kelvin Ezenwaka
Sent: Friday, May 13, 2022 12:46 PM
To: users <users@lists.kamailio.org>; sr-users-owner \
<sr-users-owner@lists.kamailio.org>
Cc: ope@tosoplimited.com; Ope Shokunbi <ope@pressone.co>
Subject: [SR-Users] Failing Audio stream on Kamailio+webrtc with integration to \
asterisk server
Summary:
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:
Webrtc client<------->Kamailio1+rtpengine<-----Asterisk---->Kamailio2<-------->Telco \
Provider
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. Asterisk 18.9.0
Kamailio 5.5.3 + RTPengine 10.4.0.0
Debian 11 bullseye
Kindly see attached below a diagram depicting the VoIP network flow and also attached \
are logs files for the webrtc client and server side.
Ezenwaka Kelvin Arinze
VOIP Engineer
Tel: (+234) 9056925668
DISCLAIMER:
The message and its attachments are for designated recipient(s) only and may contain \
privileged, proprietary and private information. If you have received it in error, \
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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> <div>
<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> </div>
<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> <div>
<div style="border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm">
<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> </div>
</div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
lang="EN-GB"><o:p> </o:p></span></p> <div>
<p style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt;background:white">
<span style="font-size:10.5pt;font-family:"Segoe \
UI",sans-serif;color:#24292F">Summary:<o:p></o:p></span></p> <p \
style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt;background:white;box-sizing: \
border-box;font-variant-ligatures: normal;font-variant-caps: normal;orphans: \
2;text-align:start;widows: 2;-webkit-text-stroke-width: \
0px;text-decoration-thickness: initial;text-decoration-style: \
initial;text-decoration-color: initial;word-spacing:0px"> <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> <p \
style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt;background:white;box-sizing: \
border-box;font-variant-ligatures: normal;font-variant-caps: normal;orphans: \
2;text-align:start;widows: 2;-webkit-text-stroke-width: \
0px;text-decoration-thickness: initial;text-decoration-style: \
initial;text-decoration-color: initial;word-spacing:0px"> <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> <p \
style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt;background:white;box-sizing: \
border-box;font-variant-ligatures: normal;font-variant-caps: normal;orphans: \
2;text-align:start;widows: 2;-webkit-text-stroke-width: \
0px;text-decoration-thickness: initial;text-decoration-style: \
initial;text-decoration-color: initial;word-spacing:0px"> <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> <div>
<p class="MsoNormal" style="margin-left:35.4pt;box-sizing: \
border-box;font-variant-ligatures: normal;font-variant-caps: normal;orphans: \
2;text-align:start;widows: 2;-webkit-text-stroke-width: \
0px;text-decoration-thickness: initial;text-decoration-style: \
initial;text-decoration-color: initial;word-spacing:0px"> <span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif">Asterisk \
18.9.0<o:p></o:p></span></p> </div>
<div>
<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> </div>
<div>
<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> </div>
<p style="mso-margin-top-alt:0cm;margin-right:0cm;margin-bottom:12.0pt;margin-left:35.4pt;background:white;box-sizing: \
border-box;font-variant-ligatures: normal;font-variant-caps: normal;orphans: \
2;text-align:start;widows: 2;-webkit-text-stroke-width: \
0px;text-decoration-thickness: initial;text-decoration-style: \
initial;text-decoration-color: initial;word-spacing:0px"> <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> <div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p> </o:p></span></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p> </o:p></span></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p> </o:p></span></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p> </o:p></span></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p> </o:p></span></p>
</div>
<div>
<div id="">
<div>
<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>
</div>
<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> \
</div> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Verdana",sans-serif"><o:p> </o:p></span></p>
</div>
</div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><o:p> </o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Arial",sans-serif;color:gray">DISCLAIMER:</span><o:p></o:p></p>
</div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Arial",sans-serif;color:gray">The \
message and its attachments are for designated recipient(s) only and may contain \
privileged, proprietary and private information. If you have received it in error, \
kindly delete it and notify the sender immediately.</span><o:p></o:p></p> </div>
<div>
<p class="MsoNormal" style="margin-left:35.4pt"><span \
style="font-size:10.0pt;font-family:"Arial",sans-serif;color:gray">PressOne \
accepts no liability for any loss or damage resulting directly and indirectly from \
the transmission of this e-mail message.</span><o:p></o:p></p> </div>
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