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:
######################### START OF LOG ########################## # Call from Grandstream to browser ################################## [root@sip ~]# ngrep -d any -qt -W byline port 5060 interface: any filter: ( port 5060 ) and (ip or ip6) ... U 2016/05/19 09:15:24.701195 192.168.100.85:5060 -> 192.168.100.159:5060 INVITE sip:1001@192.168.100.159 SIP/2.0. Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca. From: "Fernando" sip:1002@192.168.100.159;tag=b0d53bed080e1b0f. To: sip:1001@192.168.100.159. Contact: sip:1002@192.168.100.85:5060;transport=udp. Supported: replaces, timer, path. P-Early-Media: Supported. Proxy-Authorization: Digest username="1002", realm="192.168.100.159", algorithm=MD5, uri="sip:1001@192.168.100.159", nonce="Vz13SFc9dhwixIQaHqfrXSDvtNLkf+guJwNHi4A=", response="9c4f2fc2b7f179172fe9a6adf0d2f60f". Call-ID: dda078a035a57ecb@192.168.100.85. CSeq: 8700 INVITE. User-Agent: Grandstream BT200 1.2.5.3. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 385. . v=0. o=1002 8000 8001 IN IP4 192.168.100.85. s=SIP Call. c=IN IP4 192.168.100.85. t=0 0. m=audio 11022 RTP/AVP 18 8 0 3 9 2 97 101. a=sendrecv. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:9 G722/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=20. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2016/05/19 09:15:24.710083 192.168.100.159:5060 -> 192.168.100.85:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca. From: "Fernando" sip:1002@192.168.100.159;tag=b0d53bed080e1b0f. To: sip:1001@192.168.100.159;tag=56f8047e80cfcc9e90a3acc9609da1ba-9e91. Call-ID: dda078a035a57ecb@192.168.100.85. CSeq: 8700 INVITE. Server: kamailio (4.2.3 (x86_64/linux)). Content-Length: 0.
######################### END OF LOG ##########################
So the Grandstream offers a lot of codecs but will get a "Not Found" from Kamailio. Look in the other way:
######################### START OF LOG ########################## # Call from browser to Grandstream ################################## [root@sip ~]# ngrep -d any -qt -W byline port 5060 interface: any filter: ( port 5060 ) and (ip or ip6) U 2016/05/19 09:25:11.285826 192.168.100.159:5060 -> 192.168.100.85:5060 INVITE sip:1002@192.168.100.85:5060;transport=udp SIP/2.0. Record-Route: sip:192.168.100.159;r2=on;lr=on. Record-Route: sip:192.168.100.159:4443;transport=ws;r2=on;lr=on. Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0. Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318. From: "Moacir"sip:1001@my.lab;tag=MhRTswgaXENAxcDi25HJ. To: sip:1002@my.lab. 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". Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d. CSeq: 15365 INVITE. Content-Type: application/sdp. Content-Length: 1182. Max-Forwards: 69. User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04. Organization: Doubango Telecom. . v=0. o=mozilla...THIS_IS_SDPARTA-46.0.1 947803314240298800 0 IN IP4 127.0.0.1. s=Doubango Telecom - firefox. t=0 0. a=sendrecv. 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. a=ice-options:trickle. a=msid-semantic:WMS *. m=audio 61455 UDP/TLS/RTP/SAVPF 109 9 0 8. c=IN IP4 192.168.100.249. a=candidate:0 1 UDP 2122252543 192.168.100.249 61455 typ host. a=candidate:1 1 UDP 2122187007 2001:0:5ef5:79fd:24be:2fcf:fa06:dbc8 61456 typ host. a=candidate:0 2 UDP 2122252542 192.168.100.249 61457 typ host. a=candidate:1 2 UDP 2122187006 2001:0:5ef5:7 U 2016/05/19 09:25:11.348673 192.168.100.85:5060 -> 192.168.100.159:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0. Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318. Record-Route: sip:192.168.100.159;r2=on;lr=on. Record-Route: sip:192.168.100.159:4443;transport=ws;r2=on;lr=on. From: "Moacir"sip:1001@my.lab;tag=MhRTswgaXENAxcDi25HJ. To: sip:1002@my.lab;tag=9ea71e1d1f839cef. Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d. CSeq: 15365 INVITE. User-Agent: Grandstream BT200 1.2.5.3. Warning: 304 GS "Media type not available". Content-Length: 0.
######################### END OF LOG ##########################
Here the Grandstream says "Media type not available". As I am not a real SIP guy, I got no clue why does not work!
Anyway, I am using the latest RPMs from Kamailo, running it using the websocket.cfg suggested configuration, no rtpengine installed on it. At the WebRTC, I am using sipml5 configuring it not to use STUN/TURN.
Cheers! Moacir
To: sr-users@lists.sip-router.org From: miconda@gmail.com Date: Thu, 19 May 2016 06:22:57 +0200 Subject: Re: [SR-Users] Browser WebRTC transcoder
What codecs are supported by your grandstream? Isn't the g711 in the group?
Cheers,
Daniel
On 19/05/16 01:51, Moacir Ferreira wrote:
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?
Cheers,
Moacir
> To: sr-users@lists.sip-router.org
> From: rfuchs@sipwise.com
> Date: Wed, 18 May 2016 19:03:10 -0400
> Subject: Re: [SR-Users] Browser WebRTC transcoder
>
> On 18/05/16 04:57 PM, Moacir Ferreira wrote:
> > Hey Daniel,
> >
> > If you say so, you probably right... I did not try it because on the
> > sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
> >
> > /"Rtpengine does not (yet) support:/
> > //
> >
> > * /Repacketization or transcoding/
>
> This refers to translating one audio codec into another (e.g. opus to
> PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting)
> is supported.
>
> Cheers
>
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