One week ago I has configured master with sipML5. You can try reproduce. On Fri, Apr 8, 2016, 06:52 Quan Huo Sheng <quanhs@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@lists.freeswitch.org [mailto: freeswitch-users-bounces@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@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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@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@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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@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@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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <freeswitch-users-bounces@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@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@lists.freeswitch.org [ mailto:freeswitch-users-bounces@lists.freeswitch.org <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
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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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