[Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

[Quan Huo Sheng]

OK. At point of time now, I prefer SIP to Verto as Webrtc signaling as I am familiar with SIP.

Well, can help to provide workable configuration. Or troubleshooting the issue. Thanks a lot.

As understanding, CODEC NEGOTIATION DEAY is set, how does FS include SRTP-DTLS (UDP/TLS/RTP/SAVPF) in m line of SDP to be sent to Chrome endpoint.

In my case, in the SDP sent from FS to 1010 (Chrome endpoint, Registered via WSS), there is no such information but normal RTP (RTP/SAVP).

Coming back, the key issue is still CODEC NEGOTIATION ERR for WebRTC.


Regards
Smile.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, April 06, 2016 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

yes it's supported

On Tuesday, April 5, 2016, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
Hi Itola;

Sorry, same error.

Does Freeswitch support media switching (srtp-dtls) between two chrome(sip.js as signal) browsers?

Finding when FS runs in media mode:
codec causes caller side “488 not acceptable here| incompatible destination ”
                callee side: “cancel |user not registered”

Regards
Smile

From: freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');> [mailto:freeswitch-users-bounces at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces at lists.freeswitch.org');>] On Behalf Of Ítalo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users at lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','freeswitch-users at lists.freeswitch.org');>
Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP

Set media_mix_inbound_outbound_codecs=true

Ítalo Rossi
italo at freeswitch.org<javascript:_e(%7B%7D,'cvml','italo at freeswitch.org');>
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On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<javascript:_e(%7B%7D,'cvml','quanhs at stee.stengg.com');>> wrote:

Hi All;



I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp).  Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min<http://192.168.199.216/chrome/sipjs:sip-0.7.0.min>) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min<http://192.168.199.216/chrome/sip-0.7.0.min>),just following the information in book “FreeSWITCH 1.6 Cookbook”.

If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other.

But if disabling media bypass mode, call is rejected by FS.



FS log shows “[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP”. Attachment is detailed FS log file.

Chrome uses opus 111, FS uses opus 116.

Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp().



Help is needed to troubleshoot this issue.



Thanks advance.

Smile.



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