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

Thu Apr 7 10:46:49 MSD 2016

freeswitch doesn't, nor should it ever, include "UDP/TLS/RTP/SAVPF" in an sdp.  Can you please explain what exactly you think is missing from the sdp in our offer?

> On Apr 7, 2016, at 2:01 AM, Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
> 
> I am using sip.js (sip-0.7.0.min as mentioned at book “FreeSWITCH 1.6 cookbook |author: Anthony Minessale ”), not jssip.
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Thursday, April 07, 2016 1:53 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> you keep saying 
> 
> (UDP/TLS/RTP/SAVPF)
>  
> are you thinking that is supposed to be in the sip message somehow?  sounds like you are using jssip, which is known to have issues.
> 
> On Wednesday, April 6, 2016, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi;
>  
> Forgot another information. Does CODEC VP8 must be included in codec_prefs in WebRTC regardless there is not at all video involved? 
>  
>  
> Regards
> Smile
>  
> From: freeswitch-users-bounces at lists.freeswitch.org <> [mailto:freeswitch-users-bounces at lists.freeswitch.org <>] On Behalf Of Quan Huo Sheng
> Sent: Thursday, April 07, 2016 10:41 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>  
> 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 <>> 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 <mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto: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 <mailto: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>
> IRC chat.freenode.net <http://chat.freenode.net/> #freeswitch #freeswitch-dev
> Bugs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Docs? https://freeswitch.org/jira <https://freeswitch.org/jira>
> Chat? https://hipchat.freeswitch.org/gUdAgy0m6 <https://hipchat.freeswitch.org/gUdAgy0m6>
>  
>  
> 
> 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 <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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