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

[Quan Huo Sheng]

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 [mailto: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
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<mailto:italo at freeswitch.org>
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On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto: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) ->freeswitch (192.168.199.22) ->1010 (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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