[Freeswitch-users] SDP with RTP SAVPF is rejected on some SIP phones
john.smolka9 at gmail.com
Fri May 6 11:07:44 MSD 2016
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
On sip phones it is rejected with 488 Not Acceptable Here
SDP from freeswitch's INVITE looks like :
o=My-SBC 1462416600 1462416601 IN IP4 MyPublicIP.
c=IN IP4 MyPublicIP.
a=msid-semantic: WMS 21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf.
m=audio 32204 RTP/SAVPF 8 0 101 13.
a=rtcp:32204 IN IP4 MyPublicIP.
a=ssrc:590458548 msid:21zmABqHTMGtfiYcvg2nQyTyTBJpmPmf a0.
a=candidate:9311553015 1 udp 659136 MyPublicIP 32204 typ host generation 0.
a=candidate:9311553015 2 udp 659136 MyPublicIP 32204 typ host generation 0.
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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