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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Ítalo Rossi
Sent: Tuesday, April 05, 2016 8:58 PM
To: FreeSWITCH Users Help
Cc: freeswitch-users@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@freeswitch.org

IRC chat.freenode.net #freeswitch #freeswitch-dev

Bugs? https://freeswitch.org/jira

Docs? https://freeswitch.org/jira

Chat? https://hipchat.freeswitch.org/gUdAgy0m6

 



On Apr 4 2016, at 11:45 pm, Quan Huo Sheng <quanhs@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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