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<body class='hmmessage'><div dir='ltr'>Yes G711 is offered... I guess the Grandstream phone is "confused" about the way the SIP browser stack talks. I got the following logs taken on Kamailio, using ngrep:<BR> <BR>######################### START OF LOG ##########################<BR># Call from Grandstream to browser<br>##################################<br>[root@sip ~]# ngrep -d any -qt -W byline port 5060<br>interface: any<br>filter: ( port 5060 ) and (ip or ip6)<BR>...<BR>U 2016/05/19 09:15:24.701195 192.168.100.85:5060 -> 192.168.100.159:5060<br>INVITE sip:1001@192.168.100.159 SIP/2.0.<br>Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca.<br>From: "Fernando" <sip:1002@192.168.100.159>;tag=b0d53bed080e1b0f.<br>To: <sip:1001@192.168.100.159>.<br>Contact: <sip:1002@192.168.100.85:5060;transport=udp>.<br>Supported: replaces, timer, path.<br>P-Early-Media: Supported.<br>Proxy-Authorization: Digest username="1002", realm="192.168.100.159", algorithm=MD5, uri="sip:1001@192.168.100.159", nonce="Vz13SFc9dhwixIQaHqfrXSDvtNLkf+guJwNHi4A=", response="9c4f2fc2b7f179172fe9a6adf0d2f60f".<br>Call-ID: <a href="mailto:dda078a035a57ecb@192.168.100.85">dda078a035a57ecb@192.168.100.85</a>.<br>CSeq: 8700 INVITE.<br>User-Agent: Grandstream BT200 1.2.5.3.<br>Max-Forwards: 70.<br>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK.<br>Content-Type: application/sdp.<br>Content-Length: 385.<br>.<br>v=0.<br>o=1002 8000 8001 IN IP4 192.168.100.85.<br>s=SIP Call.<br>c=IN IP4 192.168.100.85.<br>t=0 0.<br>m=audio 11022 RTP/AVP 18 8 0 3 9 2 97 101.<br>a=sendrecv.<br>a=rtpmap:18 G729/8000.<br>a=rtpmap:8 PCMA/8000.<br>a=rtpmap:0 PCMU/8000.<br>a=rtpmap:3 GSM/8000.<br>a=rtpmap:9 G722/8000.<br>a=rtpmap:2 G726-32/8000.<br>a=rtpmap:97 iLBC/8000.<br>a=fmtp:97 mode=20.<br>a=ptime:20.<br>a=rtpmap:101 telephone-event/8000.<br>a=fmtp:101 0-11.<BR><br>U 2016/05/19 09:15:24.710083 192.168.100.159:5060 -> 192.168.100.85:5060<br>SIP/2.0 404 Not Found.<br>Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca.<br>From: "Fernando" <sip:1002@192.168.100.159>;tag=b0d53bed080e1b0f.<br>To: <sip:1001@192.168.100.159>;tag=56f8047e80cfcc9e90a3acc9609da1ba-9e91.<br>Call-ID: <a href="mailto:dda078a035a57ecb@192.168.100.85">dda078a035a57ecb@192.168.100.85</a>.<br>CSeq: 8700 INVITE.<br>Server: kamailio (4.2.3 (x86_64/linux)).<br>Content-Length: 0.<br><BR>######################### END OF LOG ##########################<BR> <BR>So the Grandstream offers a lot of codecs but will get a "Not Found" from Kamailio. Look in the other way:<BR> <BR>######################### START OF LOG ##########################<BR># Call from browser to Grandstream<br>##################################<br>[root@sip ~]# ngrep -d any -qt -W byline port 5060<br>interface: any<br>filter: ( port 5060 ) and (ip or ip6)<BR>U 2016/05/19 09:25:11.285826 192.168.100.159:5060 -> 192.168.100.85:5060<br>INVITE sip:1002@192.168.100.85:5060;transport=udp SIP/2.0.<br>Record-Route: <sip:192.168.100.159;r2=on;lr=on>.<br>Record-Route: <sip:192.168.100.159:4443;transport=ws;r2=on;lr=on>.<br>Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0.<br>Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318.<br>From: "Moacir"<sip:1001@my.lab>;tag=MhRTswgaXENAxcDi25HJ.<br>To: <sip:1002@my.lab>.<br>Contact: "Moacir"<sips:1001@df7jal23ls0d.invalid;alias=192.168.100.249~59318~6;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr".<br>Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d.<br>CSeq: 15365 INVITE.<br>Content-Type: application/sdp.<br>Content-Length: 1182.<br>Max-Forwards: 69.<br>User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04.<br>Organization: Doubango Telecom.<br>.<br>v=0.<br>o=mozilla...THIS_IS_SDPARTA-46.0.1 947803314240298800 0 IN IP4 127.0.0.1.<br>s=Doubango Telecom - firefox.<br>t=0 0.<br>a=sendrecv.<br>a=fingerprint:sha-256 44:F1:6D:31:F4:D6:D9:43:1D:38:0B:8E:67:1E:5F:DD:10:F4:5F:1C:4B:7E:7A:47:F8:85:C4:93:40:A7:2D:5E.<br>a=ice-options:trickle.<br>a=msid-semantic:WMS *.<br>m=audio 61455 UDP/TLS/RTP/SAVPF 109 9 0 8.<br>c=IN IP4 192.168.100.249.<br>a=candidate:0 1 UDP 2122252543 192.168.100.249 61455 typ host.<br>a=candidate:1 1 UDP 2122187007 2001:0:5ef5:79fd:24be:2fcf:fa06:dbc8 61456 typ host.<br>a=candidate:0 2 UDP 2122252542 192.168.100.249 61457 typ host.<br>a=candidate:1 2 UDP 2122187006 2001:0:5ef5:7<BR>U 2016/05/19 09:25:11.348673 192.168.100.85:5060 -> 192.168.100.159:5060<br>SIP/2.0 488 Not Acceptable Here.<br>Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0.<br>Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318.<br>Record-Route: <sip:192.168.100.159;r2=on;lr=on>.<br>Record-Route: <sip:192.168.100.159:4443;transport=ws;r2=on;lr=on>.<br>From: "Moacir"<sip:1001@my.lab>;tag=MhRTswgaXENAxcDi25HJ.<br>To: <sip:1002@my.lab>;tag=9ea71e1d1f839cef.<br>Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d.<br>CSeq: 15365 INVITE.<br>User-Agent: Grandstream BT200 1.2.5.3.<br>Warning: 304 GS "Media type not available".<br>Content-Length: 0.<br><BR>######################### END OF LOG ##########################<BR> <BR>Here the Grandstream says "Media type not available". As I am not a real SIP guy, I got no clue why does not work!<BR> <BR>Anyway, I am using the latest RPMs from Kamailo, running it using the websocket.cfg suggested configuration, <strong>no</strong> rtpengine installed on it. At the WebRTC, I am using sipml5 configuring it not to use STUN/TURN.<BR> <BR>Cheers!<BR>Moacir<BR> <BR><div><hr id="stopSpelling">To: sr-users@lists.sip-router.org<br>From: miconda@gmail.com<br>Date: Thu, 19 May 2016 06:22:57 +0200<br>Subject: Re: [SR-Users] Browser WebRTC transcoder<br><br>
What codecs are supported by your grandstream? Isn't the g711 in
the group?<BR>
Cheers,<br>
Daniel<br>
<BR>
<br>
<div class="ecxmoz-cite-prefix">On 19/05/16 01:51, Moacir Ferreira
wrote:<br>
</div>
<blockquote cite="mid:COL131-W56B6768694847AEDCC959CC8490@phx.gbl">
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<div dir="ltr">I did not dig into the problem but on my tests I
saw that my (old) Grandstream phone was refusing the call for
not having a compatible codec to talk with the offered ones by
the browser (Firefox). Being this the case, I guess I must
include a translator, and all routing logic, in between the
callers. It points to Asterisk that I would like to avoid for
now. But I guess this is not a problem that only affects me.
Someone else must have faced this before. So the question still
open: What solution would be recommended for such case?<br>
<br>
Cheers,<br>
Moacir<br>
<br>
<div>> To: <a class="ecxmoz-txt-link-abbreviated" href="mailto:sr-users@lists.sip-router.org">sr-users@lists.sip-router.org</a><br>
> From: <a class="ecxmoz-txt-link-abbreviated" href="mailto:rfuchs@sipwise.com">rfuchs@sipwise.com</a><br>
> Date: Wed, 18 May 2016 19:03:10 -0400<br>
> Subject: Re: [SR-Users] Browser WebRTC transcoder<br>
> <br>
> On 18/05/16 04:57 PM, Moacir Ferreira wrote:<br>
> > Hey Daniel,<br>
> ><br>
> > If you say so, you probably right... I did not try
it because on the<br>
> > sipwise GitHub
(<a class="ecxmoz-txt-link-freetext" href="https://github.com/sipwise/rtpengine" target="_blank">https://github.com/sipwise/rtpengine</a>) they mention:<br>
> ><br>
> > /"Rtpengine does not (yet) support:/<br>
> > //<br>
> ><br>
> > * /Repacketization or transcoding/<br>
> <br>
> This refers to translating one audio codec into another
(e.g. opus to <br>
> PCM). Translating between RTP and SRTP (i.e. encrypting
and decrypting) <br>
> is supported.<br>
> <br>
> Cheers<br>
> <br>
> _______________________________________________<br>
> SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list<br>
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<pre class="ecxmoz-signature">--
Daniel-Constantin Mierla
<a class="ecxmoz-txt-link-freetext" href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a class="ecxmoz-txt-link-freetext" href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a class="ecxmoz-txt-link-freetext" href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
Kamailio World Conference, Berlin, May 18-20, 2016 - <a class="ecxmoz-txt-link-freetext" href="http://www.kamailioworld.com" target="_blank">http://www.kamailioworld.com</a></pre>
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