Could you share your Kamailio config.<div>I now troubleshoot call to WebRTC clients, and this will help me.<br><br><div class="gmail_quote"><div dir="ltr">ср, 20 февр. 2019 г., 10:52 Adesh Pandey &lt;<a href="mailto:adesh.pandey@myoperator.co">adesh.pandey@myoperator.co</a>&gt;:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div dir="ltr"><div>Hi <br></div><div>I have configured Kamailio as dispatcher for Asterisk, if I am logging in to kamilio with webrtc can make calls with Asterisk and everything goes fine but when I try to make calls from Zoiper my asterisk box says that :</div><div><br></div><div>[2019-02-19 08:45:15] WARNING[4649][C-00000008] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio<br>[2019-02-19 08:45:15] NOTICE[4649][C-00000008] chan_sip.c: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 8000 RTP/AVP 106 9 3 111 0 8 97 110 112 98 101 100 99 102<br>[2019-02-19 08:45:15] VERBOSE[4649][C-00000008] netsock2.c: Using SIP RTP CoS mark 5<br>[2019-02-19 08:45:15] VERBOSE[4649][C-00000008] res_rtp_asterisk.c: DTLS ECDH initialized (secp256r1), faster PFS enabled</div><div><br></div><div>I have explored over the internet but nothing found, please help me to modify AVP in such a way that RTP request gets processed by Asterisk.<br></div></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Mon, Feb 18, 2019 at 7:12 PM Daniel-Constantin Mierla &lt;<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>&gt; wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">Hello,<br>
<br>
On 18.02.19 10:07, Adesh Pandey wrote:<br>
&gt; Hi Guys,<br>
&gt; I have recently setup a Kamailio server which can accept SIP or<br>
&gt; WebSocket connections but voice is not coming in  the case of SIP client.<br>
&gt;<br>
&gt; I have no idea to change avp in the incoming request, please advise<br>
&gt; how to proceed.<br>
<br>
the details you provide here doesn&#39;t give any clue of what is the real<br>
problem, where and why, so likely nobody can really help directly at<br>
this stage.<br>
<br>
I suggest to start with debug=3 in kamailio cfg and watch the logs when<br>
you do testing calls. Watch also the web browser console/diagnostic<br>
tools to see if you get any hints there.<br>
<br>
If you search on the web, you should fine some tutorials about using<br>
kamailio in webrtc -- they might be a good reference to compare with<br>
your config.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
-- <br>
Daniel-Constantin Mierla -- <a href="http://www.asipto.com" rel="noreferrer" target="_blank">www.asipto.com</a><br>
<a href="http://www.twitter.com/miconda" rel="noreferrer" target="_blank">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" rel="noreferrer" target="_blank">www.linkedin.com/in/miconda</a><br>
Kamailio World Conference - May 6-8, 2019 -- <a href="http://www.kamailioworld.com" rel="noreferrer" target="_blank">www.kamailioworld.com</a><br>
Kamailio Advanced Training - Mar 4-6, 2019 in Berlin; Mar 25-27, 2019, in Washington, DC, USA -- <a href="http://www.asipto.com" rel="noreferrer" target="_blank">www.asipto.com</a><br>
<br>
</blockquote></div>
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