[Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
Michael Jerris
mike at jerris.com
Thu Apr 7 18:40:28 MSD 2016
you can search jira and the mailing list.. there were many issues. I guess its possible they came back and fixed them, but last i looked they went many months with no response to the bugs.
> On Apr 7, 2016, at 3:17 AM, saurabh verrma <saurabhkv01 at gmail.com> wrote:
>
> Hi Michael,
>
> I’ve been working on having FreeSWITCH interact with JsSIP for WebRTC calling. You’ve mentioned here that JsSIP is known to be having issues.
>
> Could you please point me to some link which gives details about the issue or do the favour of telling me what are the known issues in FreeSWITCH + JsSIP.
>
> It would be appreciable.
>
> On Thu, Apr 7, 2016 at 12:35 PM, Quan Huo Sheng <quanhs at stee.stengg.com <mailto:quanhs at stee.stengg.com>> wrote:
> Hi Michael;
>
>
>
> Key exchange via sdp (SDES) is no longer supported by chrome. Chrome only supports SRTP-DTLS (UDP/TLS/RTP/SAVPF in m line of SDP) in case SIP used as WebRTC signaling.
>
>
>
> You can see SDP from chrome (+sipjs) for this in previous attachments.
>
>
>
> If set inbound-bypass-media=true, then chrome caller can talk with chrome callee using opus without any issue.
>
>
>
>
>
> 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 Michael Jerris
> Sent: Thursday, April 07, 2016 2:47 PM
>
>
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
>
>
>
> 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 <mailto: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> [mailto: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> [mailto: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> [mailto: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 <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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> _________________________________________________________________________
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>
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>
> --
> With Warm Regards:
> Saurabh Kumar Verma
> _________________________________________________________________________
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>
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