Freeswitch and webRTC: media rejected with 488

I can register from my webclient to my freeswitch. But, when I try to make call the call gets rejected with 488 not acceptable here. From freeswitch console log im getting.

2014-07-22 22:03:59.673585 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice@192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION

I added < action application="export" data="rtp_secure_media=true" /> with my extension; but no luck.

below is the SDP of my INVITE

v=0
o=Mozilla-SIPUA-31.0 26508 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:13497e25
a=ice-pwd:515d61f08d909117e022674f3dce748e
a=fingerprint:sha-256 2E:CF:7E:8F:EC:1A:F4:B1:D3:CF:39:C3:8A:A0:D0:53:B3:46:00:D0:93:46:53:29:AB:B7:03:83:39:FB:23:32
m=audio 55760 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 184.69.59.132
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.16.1.188 55760 typ host
a=candidate:1 1 UDP 1692467199 184.69.59.132 55760 typ srflx raddr 172.16.1.188 rport 55760
a=candidate:5 1 UDP 2128543999 192.168.56.1 55761 typ host
a=candidate:10 1 UDP 2128478463 192.168.232.1 55762 typ host
a=candidate:15 1 UDP 2128412927 192.168.146.1 55763 typ host
a=candidate:0 2 UDP 2128609534 172.16.1.188 55764 typ host
a=candidate:1 2 UDP 1692467198 184.69.59.132 55764 typ srflx raddr 172.16.1.188 rport 55764
a=candidate:5 2 UDP 2128543998 192.168.56.1 55765 typ host
a=candidate:10 2 UDP 2128478462 192.168.232.1 55766 typ host
a=candidate:15 2 UDP 2128412926 192.168.146.1 55767 typ host
a=rtcp-mux

Below is my codec lists from freeswitch. I dont have opus installed, but I do have G711 ulaw and alaw

show codecs
type,name,ikey
codec,ADPCM (IMA),mod_voipcodecs
codec,AMR,mod_amr
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_voipcodecs
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_voipcodecs
codec,G.726 16k (AAL2),mod_voipcodecs
codec,G.726 24k,mod_voipcodecs
codec,G.726 24k (AAL2),mod_voipcodecs
codec,G.726 32k,mod_voipcodecs
codec,G.726 32k (AAL2),mod_voipcodecs
codec,G.726 40k,mod_voipcodecs
codec,G.726 40k (AAL2),mod_voipcodecs
codec,G.729,mod_g729
codec,GSM,mod_voipcodecs
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_voipcodecs
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,Polycom(R) G722.1/G722.1C,mod_siren
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,iLBC,mod_ilbc

What can be the issue?

回答1:

Based on the logs it could also be a compatibilty issue with the m line that has all the transport protocols listed together.

UDP/TLS/RTP/SAVPF. This can be subject to compatibility issues as mentioned in these threads. Maybe you could try to restrict it to the simpler form and try it if possible.

https://code.google.com/p/webrtc/issues/detail?id=2796

http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097617.html

回答2:

Most often a 488 rejection is caused by codec mismatch. Please check the FS and the WebRTC settings. Usually WebRTC uses Opus so you need to make sure that selected in the FS Config [if possible].

A PCAP of the Issue or screen shots of the INVITE and the 488 can help narrow down the problem further.