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

Thu Apr 7 11:17:56 MSD 2016

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>
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] *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> 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
> <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>
> 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
> <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
> <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
> <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
>
> 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 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.
>
>
>
>
> [This e-mail is confidential and may be privileged. If you are not the
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>
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>
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>
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-- 
*With Warm Regards:*
*Saurabh Kumar Verma*
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