[Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
Quan Huo Sheng
quanhs at stee.stengg.com
Wed Apr 6 07:35:39 MSD 2016
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”
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
italo at freeswitch.org<mailto:italo at freeswitch.org>
IRC chat.freenode.net #freeswitch #freeswitch-dev
On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs at stee.stengg.com<mailto:quanhs at stee.stengg.com>> wrote:
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.
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