[Freeswitch-users] SDP with RTP SAVPF is rejected on some SIP phones

[John Smolka]

nobody uses webrtc and classic calls with freeswitch as a media server and
kamailio or opensips as a sip proxy ?
I found a solution with rtpengine, but I want to use freeswitch as a media
server.



2016-05-06 9:07 GMT+02:00 John Smolka <john.smolka9 at gmail.com>:

> Hello All,
>
> I am working with webrtc and I am having problems with incoming audio.
> I use :
> - Kamailio as a SIP and Websocket proxy
> - Freeswitch as a media server.
>
> UA/WebRTC <--> Kamailio <--> Freeswitch
>
> For outgoing calls (webrtc->freeswitch) everything works well, but for
> incoming call(freeswitch->webrtc), my webrtc client was complaining about
> missing ICE candidates.
> In my dialplan I added a new line with:
>
> <action application="export" data="media_webrtc=true" />
>
> since now, RTP/AVP and RTP/SAVP was changed to RTP/SAVPF
>
> Now, I can receive calls on my webrtc client but it does not work on some
> sip phones.
> On sip phones it is rejected with 488 Not Acceptable Here
>
> SDP from freeswitch's INVITE looks like :
>
> v=0.
> o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP.
> s=My-SBC.
> c=IN IP4 MyPublicIP.
> t=0 0.
> a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf.
> m=audio 32204 RTP/SAVPF 8 0 101 13.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fingerprint:sha-256 B8:7C:42:58:26:FE:30:27:DB:F5:
> 71:21:11:96:AF:0B:DD:6E:79:9A:AC:FF:86:69:89:14:88:70:AA:EE:4A:EB.
> a=setup:actpass.
> a=rtcp-mux.
> a=rtcp:32204 IN IP4 MyPublicIP.
> a=ssrc:590458548 cname:GLr7dIFWlcfql4sH.
> a=ssrc:590458548 msid:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf a0.
> a=ssrc:590458548 mslabel:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf.
> a=ssrc:590458548 label:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmfa0.
> a=ice-ufrag:akVRJFgs6QHjvUo2.
> a=ice-pwd:vG0zewmpyrGKmZhm924Jxa1u.
> a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation
> 0.
> a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation
> 0.
> a=ptime:20.
>
> Maybe I don't understand this correctly, but is that possible to add
> SAVPF, AVP and SAVP into a INVITES's SDP so I can remove them on kamailio
> based on transport towards client (TLS,UDP,TCP,WSS) ?
>
> I have tested it on the latest freeswitch 1.4 and 1.6 versions.
>
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