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

Fri Apr 8 11:17:13 MSD 2016

All call with media transcoding enabled. In WebRTC case OPUS <-> G711a

On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs at stee.stengg.com> wrote:

> what is setting of inbound-bypass-media and inbound-proxy-media in your
> case?
>
>
>
>
>
> Regards
>
> Smile
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sergey
> Safarov
> *Sent:* Friday, April 08, 2016 12:00 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> One week ago I has configured master with sipML5.
> You can try reproduce.
>
>
>
> On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs at stee.stengg.com> wrote:
>
> Hi Michael;
>
>
>
> Same complaint at mod_sofia.c 2299.
>
> Mod_sofia.c: 2299 (sofia_media_tech_media(tech_pvt, r_sdp) !=
> SWITCH_STATUS_SUCCESS).
>
> Switch_core_media.c :3573: switch_core_media_negotiation_sdp().
>
>
>
> Eval ${webrtc_enable_dtls}, Eval ${media_webrtc}, Eval ${rtp_use_dtls},
> Eval ${rtp_secure_media}, all return true.
>
>
>
> Does anyone successfully set up this WebRTC demo (excluding video) using
> media mode as described by cookbook.
>
>
>
>
>
> 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 10:17 PM
>
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC
> NEGOTIATION ERROR. SDP
>
>
>
> media_webrtc=true
>
>
>
>
>
> On Apr 7, 2016, at 3:05 AM, 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
> <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.
>
>
>
>
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> _________________________________________________________________________
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>
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> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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