webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
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 intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
Set media_mix_inbound_outbound_codecs=true Ítalo Rossi italo@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@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 intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] \---ST Electronics Group---
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] 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<mailto:italo@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@stee.stengg.com<mailto: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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
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 <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','italo@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@stee.stengg.com <javascript:_e(%7B%7D,'cvml','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.
[This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
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] 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<mailto: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<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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<javascript:_e(%7B%7D,'cvml','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<javascript:_e(%7B%7D,'cvml','italo@freeswitch.org');> IRC chat.freenode.net<http://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@stee.stengg.com<javascript:_e(%7B%7D,'cvml','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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
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] 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> [mailto: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<mailto: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<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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<javascript:_e(%7B%7D,'cvml','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<javascript:_e(%7B%7D,'cvml','italo@freeswitch.org');> IRC chat.freenode.net<http://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@stee.stengg.com<javascript:_e(%7B%7D,'cvml','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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
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 <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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] *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
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@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 intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
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] 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<mailto: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<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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<javascript:_e(%7B%7D,'cvml','freeswitch-users-bounces@lists.freeswitch.org');> [mailto:freeswitch-users-bounces@lists.freeswitch.org<javascript:_e(%7B%7D,'cvml','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<javascript:_e(%7B%7D,'cvml','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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
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] 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 <mailto: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 <>] 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 <>] 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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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 <mailto:italo@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@stee.stengg.com <mailto: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 <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.
[This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
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] 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<mailto: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> [mailto: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<mailto: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> [mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/>
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@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] *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
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@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 intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- *With Warm Regards:* *Saurabh Kumar Verma*
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@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@stee.stengg.com <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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 <mailto:italo@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@stee.stengg.com <mailto: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 <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.
[This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
-- With Warm Regards: Saurabh Kumar Verma _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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] 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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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 <mailto:italo@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@stee.stengg.com <mailto: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 <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.
[This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
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<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/>
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
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@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 intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
what is setting of inbound-bypass-media and inbound-proxy-media in your case? Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@stee.stengg.com<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
All call with media transcoding enabled. In WebRTC case OPUS <-> G711a On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs@stee.stengg.com> wrote:
what is setting of inbound-bypass-media and inbound-proxy-media in your case?
Regards
Smile
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@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@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
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@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 intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Good. Can you share your scenario ? Chrome (sipML5) ->FS (1.6.5-64bit Media mode) ->Chrome (sipML5). Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Friday, April 08, 2016 3:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP All call with media transcoding enabled. In WebRTC case OPUS <-> G711a On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs@stee.stengg.com<mailto:quanhs@stee.stengg.com>> wrote: what is setting of inbound-bypass-media and inbound-proxy-media in your case? Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@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@stee.stengg.com<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log.
On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs@stee.stengg.com> wrote:
Good. Can you share your scenario ?
Chrome (sipML5) ->FS (1.6.5-64bit Media mode) ->Chrome (sipML5).
Regards Smile
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Friday, April 08, 2016 3:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
All call with media transcoding enabled. In WebRTC case OPUS <-> G711a
On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs@stee.stengg.com <mailto:quanhs@stee.stengg.com>> wrote: what is setting of inbound-bypass-media and inbound-proxy-media in your case?
Regards Smile
From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@stee.stengg.com <mailto: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> [mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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 <mailto:italo@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@stee.stengg.com <mailto: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 <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.
[This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Because existing SIP endpoint doesn’t support SRTP-DTLS, the transcoding (SRTP-DTLS and SRTP-SDES)is required to bridge RTP stream between WebRTC browser and SIP endpoint. 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP: 1709c7c4-fa17-11e5-9184-9d31ebce310e v=0 1709c7c4-fa17-11e5-9184-9d31ebce310e o=- 2269318439683772682 2 IN IP4 127.0.0.1 1709c7c4-fa17-11e5-9184-9d31ebce310e s=- 1709c7c4-fa17-11e5-9184-9d31ebce310e t=0 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=group:BUNDLE audio 1709c7c4-fa17-11e5-9184-9d31ebce310e a=msid-semantic: WMS rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 1709c7c4-fa17-11e5-9184-9d31ebce310e m=audio 1107 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 1709c7c4-fa17-11e5-9184-9d31ebce310e c=IN IP4 192.168.199.216 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:111 opus/48000/2 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:103 ISAC/16000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:104 ISAC/32000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:9 G722/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:0 PCMU/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:8 PCMA/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:106 CN/32000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:105 CN/16000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:13 CN/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:126 telephone-event/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp:1110 IN IP4 192.168.199.216 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 1 udp 2122260223 192.168.199.216 1107 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 1 udp 2122194687 10.20.102.216 1108 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 1 udp 2122129151 10.10.13.216 1109 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 2 udp 2122260222 192.168.199.216 1110 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 2 udp 2122194686 10.20.102.216 1111 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 2 udp 2122129150 10.10.13.216 1112 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 1 tcp 1518280447 192.168.199.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 1 tcp 1518214911 10.20.102.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 1 tcp 1518149375 10.10.13.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 2 tcp 1518280446 192.168.199.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 2 tcp 1518214910 10.20.102.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 2 tcp 1518149374 10.10.13.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-ufrag:4PWOC8zOrgkF0JcH 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-pwd:KoIliPYZgGkk+b6K7bJDPVLK 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fingerprint:sha-256 18:9D:FC:F1:74:38:63:AE:F1:F8:F0:26:F5:83:A8:41:97:53:67:F5:35:FB:7E:F3:06:BF:D5:71:FD:A2:F6:38 1709c7c4-fa17-11e5-9184-9d31ebce310e a=setup:actpass 1709c7c4-fa17-11e5-9184-9d31ebce310e a=mid:audio 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp-mux 1709c7c4-fa17-11e5-9184-9d31ebce310e a=maxptime:60 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 cname:mrCkzHA0WH9Dd+8n 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 msid:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 62a659a4-ef68-469a-871c-0e61ef3dd211 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 mslabel:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 label:62a659a4-ef68-469a-871c-0e61ef3dd211 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [NOTICE] switch_channel.c:3501 Hangup sofia/internal/anonymous@192.168.199.22 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:1217 sofia/internal/anonymous@192.168.199.22 Media Establishment Failed. ############################################### B leg: 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.471064 [DEBUG] mod_sofia.c:88 sofia/internal/5shm6jpu@tibvb7p7p6od.invalid SOFIA INIT 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1228 sip:5shm6jpu@192.168.199.216:1106;transport=wss Setting proxy route to sofia/internal/5shm6jpu@tibvb7p7p6od.invalid 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1257 sofia/internal/5shm6jpu@tibvb7p7p6od.invalid sending invite version: 1.6.5 64bit 1d260ca8-fa17-11e5-91a9-9d31ebce310e Local SDP: 1d260ca8-fa17-11e5-91a9-9d31ebce310e v=0 1d260ca8-fa17-11e5-91a9-9d31ebce310e o=FreeSWITCH 1459715932 1459715933 IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e s=FreeSWITCH 1d260ca8-fa17-11e5-91a9-9d31ebce310e c=IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e t=0 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=msid-semantic: WMS V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp 1d260ca8-fa17-11e5-91a9-9d31ebce310e m=audio 25356 RTP/SAVPF 111 8 102 101 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:111 opus/48000/2 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:8 PCMA/8000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:102 telephone-event/48000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:101 telephone-event/8000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fingerprint:sha-256 E7:50:3F:56:DE:21:E5:17:DA:F7:3A:3F:28:F8:BC:2A:C6:E7:E0:67:5C:6C:61:39:00:6D:99:DE:A4:A2:8A:80 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=setup:actpass 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp-mux 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp:25356 IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 cname:OFXIzFBV4sOBkyJG 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 msid:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp a0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 mslabel:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 label:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzpa0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-ufrag:UUr7xgAHzqT3ejiT 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-pwd:O6reBS0vbEXl3xRNh8KFfbcE 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 1 udp 659136 192.168.199.22 25356 typ host generation 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 2 udp 659136 192.168.199.22 25356 typ host generation 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ptime:20 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=sendrecv @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ Profile: internal 2016-04-04 11:17:29.618618 [DEBUG] sofia.c:4236 debug [9] 2016-04-04 11:17:29.618624 [DEBUG] sofia.c:4236 sip-trace [no] 2016-04-04 11:17:29.618627 [DEBUG] sofia.c:4236 sip-capture [no] 2016-04-04 11:17:29.618630 [DEBUG] sofia.c:4236 watchdog-enabled [no] 2016-04-04 11:17:29.618634 [DEBUG] sofia.c:4236 watchdog-step-timeout [30000] 2016-04-04 11:17:29.618638 [DEBUG] sofia.c:4236 watchdog-event-timeout [30000] 2016-04-04 11:17:29.618642 [DEBUG] sofia.c:4236 log-auth-failures [true] 2016-04-04 11:17:29.618645 [DEBUG] sofia.c:4236 forward-unsolicited-mwi-notify [false] 2016-04-04 11:17:29.618648 [DEBUG] sofia.c:4236 context [public] 2016-04-04 11:17:29.618654 [DEBUG] sofia.c:4236 rfc2833-pt [101] 2016-04-04 11:17:29.618657 [DEBUG] sofia.c:4236 sip-port [5062] 2016-04-04 11:17:29.618661 [DEBUG] sofia.c:4236 dialplan [XML] 2016-04-04 11:17:29.618667 [DEBUG] sofia.c:4236 dtmf-duration [2000] 2016-04-04 11:17:29.618677 [DEBUG] sofia.c:4236 inbound-codec-prefs [OPUS] 2016-04-04 11:17:29.618684 [DEBUG] sofia.c:4236 outbound-codec-prefs [OPUS] 2016-04-04 11:17:29.618690 [DEBUG] sofia.c:4236 rtp-timer-name [soft] 2016-04-04 11:17:29.618694 [DEBUG] sofia.c:4236 rtp-ip [192.168.199.22] 2016-04-04 11:17:29.618698 [DEBUG] sofia.c:4236 sip-ip [192.168.199.22] 2016-04-04 11:17:29.618702 [DEBUG] sofia.c:4236 hold-music [local_stream://moh] 2016-04-04 11:17:29.618720 [DEBUG] sofia.c:4236 apply-nat-acl [nat.auto] 2016-04-04 11:17:29.618734 [ERR] sofia.c:5185 Not adding acl nat.auto because it's the local network 2016-04-04 11:17:29.618739 [DEBUG] sofia.c:4236 apply-inbound-acl [domains] 2016-04-04 11:17:29.618745 [DEBUG] sofia.c:4236 local-network-acl [localnet.auto] 2016-04-04 11:17:29.618749 [DEBUG] sofia.c:4236 record-path [/usr/local/freeswitch/recordings] 2016-04-04 11:17:29.618754 [DEBUG] sofia.c:4236 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2016-04-04 11:17:29.618763 [DEBUG] sofia.c:4236 manage-presence [true] 2016-04-04 11:17:29.618768 [DEBUG] sofia.c:4236 presence-hosts [192.168.199.22,192.168.199.22] 2016-04-04 11:17:29.618774 [DEBUG] sofia.c:4236 presence-privacy [false] 2016-04-04 11:17:29.618778 [DEBUG] sofia.c:4236 inbound-codec-negotiation [generous] 2016-04-04 11:17:29.618783 [DEBUG] sofia.c:4236 tls [true] 2016-04-04 11:17:29.618791 [DEBUG] sofia.c:4236 tls-only [false] 2016-04-04 11:17:29.618798 [DEBUG] sofia.c:4236 tls-bind-params [transport=tls] 2016-04-04 11:17:29.618804 [DEBUG] sofia.c:4236 tls-sip-port [5063] 2016-04-04 11:17:29.618810 [DEBUG] sofia.c:4236 tls-passphrase [] 2016-04-04 11:17:29.618816 [DEBUG] sofia.c:4236 tls-verify-date [true] 2016-04-04 11:17:29.618822 [DEBUG] sofia.c:4236 tls-verify-policy [none] 2016-04-04 11:17:29.618829 [ERR] sofia_glue.c:329 Invalid tls-verify-policy value: none 2016-04-04 11:17:29.618835 [DEBUG] sofia.c:4236 tls-verify-depth [2] 2016-04-04 11:17:29.618844 [DEBUG] sofia.c:4236 tls-verify-in-subjects [] 2016-04-04 11:17:29.618850 [DEBUG] sofia.c:4236 tls-version [tlsv1,tlsv1.1,tlsv1.2] 2016-04-04 11:17:29.618860 [DEBUG] sofia.c:4236 tls-ciphers [ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH] 2016-04-04 11:17:29.618868 [DEBUG] sofia.c:4236 inbound-late-negotiation [true] 2016-04-04 11:17:29.618872 [DEBUG] sofia.c:4236 inbound-zrtp-passthru [true] 2016-04-04 11:17:29.618876 [DEBUG] sofia.c:4236 nonce-ttl [60] 2016-04-04 11:17:29.618881 [DEBUG] sofia.c:4236 auth-calls [true] 2016-04-04 11:17:29.618887 [DEBUG] sofia.c:4236 inbound-reg-force-matching-username [true] 2016-04-04 11:17:29.618893 [DEBUG] sofia.c:4236 auth-all-packets [false] 2016-04-04 11:17:29.618898 [DEBUG] sofia.c:4236 ext-rtp-ip [auto-nat] 2016-04-04 11:17:29.618902 [DEBUG] sofia.c:4236 ext-sip-ip [auto-nat] 2016-04-04 11:17:29.618908 [DEBUG] sofia.c:4236 rtp-timeout-sec [300] 2016-04-04 11:17:29.618913 [DEBUG] sofia.c:4236 rtp-hold-timeout-sec [1800] 2016-04-04 11:17:29.618917 [DEBUG] sofia.c:4236 force-register-domain [192.168.199.22] 2016-04-04 11:17:29.618922 [DEBUG] sofia.c:4236 force-subscription-domain [192.168.199.22] 2016-04-04 11:17:29.618927 [DEBUG] sofia.c:4236 force-register-db-domain [192.168.199.22] 2016-04-04 11:17:29.618933 [DEBUG] sofia.c:4236 ws-binding [:5066] 2016-04-04 11:17:29.618939 [DEBUG] sofia.c:4236 wss-binding [:7443] 2016-04-04 11:17:29.618945 [DEBUG] sofia.c:4236 challenge-realm [auto_from] 2016-04-04 11:17:29.618953 [INFO] sofia.c:5513 Setting MAX Auth Validity to 0 Attempts 2016-04-04 11:17:29.619091 [NOTICE] sofia.c:5680 Started Profile internal [sofia_reg_internal] 2016-04-04 11:24:00.891443 [DEBUG] switch_loadable_module.c:735 Chat Thread Started 2016-04-04 11:24:00.891453 [INFO] switch_core.c:2418 FreeSWITCH Version 1.6.5~64bit ( 64bit) The detailed log is attached in previous emails. Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, April 08, 2016 11:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log. On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs@stee.stengg.com<mailto:quanhs@stee.stengg.com>> wrote: Good. Can you share your scenario ? Chrome (sipML5) ->FS (1.6.5-64bit Media mode) ->Chrome (sipML5). Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Friday, April 08, 2016 3:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP All call with media transcoding enabled. In WebRTC case OPUS <-> G711a On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs@stee.stengg.com<mailto:quanhs@stee.stengg.com>> wrote: what is setting of inbound-bypass-media and inbound-proxy-media in your case? Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@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@stee.stengg.com<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org
This is never attempting to call the second endpoint: 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] It then tries to negotiate the ice session with freeswitch, as it is trying to answer the call locally and send it to voicemail. then we have: 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:3380 sofia/internal/anonymous@192.168.199.22 no suitable candidates found. 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:4473 sofia/internal/anonymous@192.168.199.22 Set 2833 dtmf send payload to 126 recv payload to 126 the issue is your box is on a rfc1918 ip, and the default candidate acl's do not consider those valid. You will need to adjust the acl if you plan on having freeswitch as a server answering webrtc calls from private address space.
On Apr 10, 2016, at 10:11 PM, Quan Huo Sheng <quanhs@stee.stengg.com> wrote:
Because existing SIP endpoint doesn’t support SRTP-DTLS, the transcoding (SRTP-DTLS and SRTP-SDES)is required to bridge RTP stream between WebRTC browser and SIP endpoint.
1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP: 1709c7c4-fa17-11e5-9184-9d31ebce310e v=0 1709c7c4-fa17-11e5-9184-9d31ebce310e o=- 2269318439683772682 2 IN IP4 127.0.0.1 1709c7c4-fa17-11e5-9184-9d31ebce310e s=- 1709c7c4-fa17-11e5-9184-9d31ebce310e t=0 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=group:BUNDLE audio 1709c7c4-fa17-11e5-9184-9d31ebce310e a=msid-semantic: WMS rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 1709c7c4-fa17-11e5-9184-9d31ebce310e m=audio 1107 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 1709c7c4-fa17-11e5-9184-9d31ebce310e c=IN IP4 192.168.199.216 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:111 opus/48000/2 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:103 ISAC/16000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:104 ISAC/32000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:9 G722/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:0 PCMU/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:8 PCMA/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:106 CN/32000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:105 CN/16000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:13 CN/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:126 telephone-event/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp:1110 IN IP4 192.168.199.216 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 1 udp 2122260223 192.168.199.216 1107 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 1 udp 2122194687 10.20.102.216 1108 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 1 udp 2122129151 10.10.13.216 1109 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 2 udp 2122260222 192.168.199.216 1110 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 2 udp 2122194686 10.20.102.216 1111 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 2 udp 2122129150 10.10.13.216 1112 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 1 tcp 1518280447 192.168.199.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 1 tcp 1518214911 10.20.102.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 1 tcp 1518149375 10.10.13.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 2 tcp 1518280446 192.168.199.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 2 tcp 1518214910 10.20.102.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 2 tcp 1518149374 10.10.13.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-ufrag:4PWOC8zOrgkF0JcH 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-pwd:KoIliPYZgGkk+b6K7bJDPVLK 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fingerprint:sha-256 18:9D:FC:F1:74:38:63:AE:F1:F8:F0:26:F5:83:A8:41:97:53:67:F5:35:FB:7E:F3:06:BF:D5:71:FD:A2:F6:38 1709c7c4-fa17-11e5-9184-9d31ebce310e a=setup:actpass 1709c7c4-fa17-11e5-9184-9d31ebce310e a=mid:audio 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time <http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time> 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp-mux 1709c7c4-fa17-11e5-9184-9d31ebce310e a=maxptime:60 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 cname:mrCkzHA0WH9Dd+8n 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 msid:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 62a659a4-ef68-469a-871c-0e61ef3dd211 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 mslabel:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 label:62a659a4-ef68-469a-871c-0e61ef3dd211
1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [NOTICE] switch_channel.c:3501 Hangup sofia/internal/anonymous@192.168.199.22 <mailto:sofia/internal/anonymous@192.168.199.22> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:1217 sofia/internal/anonymous@192.168.199.22 <mailto:sofia/internal/anonymous@192.168.199.22> Media Establishment Failed.
############################################### B leg:
1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.471064 [DEBUG] mod_sofia.c:88 sofia/internal/5shm6jpu@tibvb7p7p6od.invalid <mailto:sofia/internal/5shm6jpu@tibvb7p7p6od.invalid> SOFIA INIT 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1228 sip:5shm6jpu@192.168.199.216:1106;transport=wss <sip:5shm6jpu@192.168.199.216:1106;transport=wss> Setting proxy route to sofia/internal/5shm6jpu@tibvb7p7p6od.invalid <mailto:sofia/internal/5shm6jpu@tibvb7p7p6od.invalid> 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1257 sofia/internal/5shm6jpu@tibvb7p7p6od.invalid <mailto:sofia/internal/5shm6jpu@tibvb7p7p6od.invalid> sending invite version: 1.6.5 64bit 1d260ca8-fa17-11e5-91a9-9d31ebce310e Local SDP: 1d260ca8-fa17-11e5-91a9-9d31ebce310e v=0 1d260ca8-fa17-11e5-91a9-9d31ebce310e o=FreeSWITCH 1459715932 1459715933 IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e s=FreeSWITCH 1d260ca8-fa17-11e5-91a9-9d31ebce310e c=IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e t=0 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=msid-semantic: WMS V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp 1d260ca8-fa17-11e5-91a9-9d31ebce310e m=audio 25356 RTP/SAVPF 111 8 102 101 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:111 opus/48000/2 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:8 PCMA/8000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:102 telephone-event/48000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:101 telephone-event/8000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fingerprint:sha-256 E7:50:3F:56:DE:21:E5:17:DA:F7:3A:3F:28:F8:BC:2A:C6:E7:E0:67:5C:6C:61:39:00:6D:99:DE:A4:A2:8A:80 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=setup:actpass 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp-mux 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp:25356 IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 cname:OFXIzFBV4sOBkyJG 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 msid:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp a0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 mslabel:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 label:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzpa0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-ufrag:UUr7xgAHzqT3ejiT 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-pwd:O6reBS0vbEXl3xRNh8KFfbcE 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 1 udp 659136 192.168.199.22 25356 typ host generation 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 2 udp 659136 192.168.199.22 25356 typ host generation 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ptime:20 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=sendrecv
@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ Profile: internal
2016-04-04 11:17:29.618618 [DEBUG] sofia.c:4236 debug [9] 2016-04-04 11:17:29.618624 [DEBUG] sofia.c:4236 sip-trace [no] 2016-04-04 11:17:29.618627 [DEBUG] sofia.c:4236 sip-capture [no] 2016-04-04 11:17:29.618630 [DEBUG] sofia.c:4236 watchdog-enabled [no] 2016-04-04 11:17:29.618634 [DEBUG] sofia.c:4236 watchdog-step-timeout [30000] 2016-04-04 11:17:29.618638 [DEBUG] sofia.c:4236 watchdog-event-timeout [30000] 2016-04-04 11:17:29.618642 [DEBUG] sofia.c:4236 log-auth-failures [true] 2016-04-04 11:17:29.618645 [DEBUG] sofia.c:4236 forward-unsolicited-mwi-notify [false] 2016-04-04 11:17:29.618648 [DEBUG] sofia.c:4236 context [public] 2016-04-04 11:17:29.618654 [DEBUG] sofia.c:4236 rfc2833-pt [101] 2016-04-04 11:17:29.618657 [DEBUG] sofia.c:4236 sip-port [5062] 2016-04-04 11:17:29.618661 [DEBUG] sofia.c:4236 dialplan [XML] 2016-04-04 11:17:29.618667 [DEBUG] sofia.c:4236 dtmf-duration [2000] 2016-04-04 11:17:29.618677 [DEBUG] sofia.c:4236 inbound-codec-prefs [OPUS] 2016-04-04 11:17:29.618684 [DEBUG] sofia.c:4236 outbound-codec-prefs [OPUS] 2016-04-04 11:17:29.618690 [DEBUG] sofia.c:4236 rtp-timer-name [soft] 2016-04-04 11:17:29.618694 [DEBUG] sofia.c:4236 rtp-ip [192.168.199.22] 2016-04-04 11:17:29.618698 [DEBUG] sofia.c:4236 sip-ip [192.168.199.22] 2016-04-04 11:17:29.618702 [DEBUG] sofia.c:4236 hold-music [local_stream://moh] 2016-04-04 11:17:29.618720 [DEBUG] sofia.c:4236 apply-nat-acl [nat.auto] 2016-04-04 11:17:29.618734 [ERR] sofia.c:5185 Not adding acl nat.auto because it's the local network 2016-04-04 11:17:29.618739 [DEBUG] sofia.c:4236 apply-inbound-acl [domains] 2016-04-04 11:17:29.618745 [DEBUG] sofia.c:4236 local-network-acl [localnet.auto] 2016-04-04 11:17:29.618749 [DEBUG] sofia.c:4236 record-path [/usr/local/freeswitch/recordings] 2016-04-04 11:17:29.618754 [DEBUG] sofia.c:4236 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2016-04-04 11:17:29.618763 [DEBUG] sofia.c:4236 manage-presence [true] 2016-04-04 11:17:29.618768 [DEBUG] sofia.c:4236 presence-hosts [192.168.199.22,192.168.199.22] 2016-04-04 11:17:29.618774 [DEBUG] sofia.c:4236 presence-privacy [false] 2016-04-04 11:17:29.618778 [DEBUG] sofia.c:4236 inbound-codec-negotiation [generous] 2016-04-04 11:17:29.618783 [DEBUG] sofia.c:4236 tls [true] 2016-04-04 11:17:29.618791 [DEBUG] sofia.c:4236 tls-only [false] 2016-04-04 11:17:29.618798 [DEBUG] sofia.c:4236 tls-bind-params [transport=tls] 2016-04-04 11:17:29.618804 [DEBUG] sofia.c:4236 tls-sip-port [5063] 2016-04-04 11:17:29.618810 [DEBUG] sofia.c:4236 tls-passphrase [] 2016-04-04 11:17:29.618816 [DEBUG] sofia.c:4236 tls-verify-date [true] 2016-04-04 11:17:29.618822 [DEBUG] sofia.c:4236 tls-verify-policy [none] 2016-04-04 11:17:29.618829 [ERR] sofia_glue.c:329 Invalid tls-verify-policy value: none 2016-04-04 11:17:29.618835 [DEBUG] sofia.c:4236 tls-verify-depth [2] 2016-04-04 11:17:29.618844 [DEBUG] sofia.c:4236 tls-verify-in-subjects [] 2016-04-04 11:17:29.618850 [DEBUG] sofia.c:4236 tls-version [tlsv1,tlsv1.1,tlsv1.2] 2016-04-04 11:17:29.618860 [DEBUG] sofia.c:4236 tls-ciphers [ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH] 2016-04-04 11:17:29.618868 [DEBUG] sofia.c:4236 inbound-late-negotiation [true] 2016-04-04 11:17:29.618872 [DEBUG] sofia.c:4236 inbound-zrtp-passthru [true] 2016-04-04 11:17:29.618876 [DEBUG] sofia.c:4236 nonce-ttl [60] 2016-04-04 11:17:29.618881 [DEBUG] sofia.c:4236 auth-calls [true] 2016-04-04 11:17:29.618887 [DEBUG] sofia.c:4236 inbound-reg-force-matching-username [true] 2016-04-04 11:17:29.618893 [DEBUG] sofia.c:4236 auth-all-packets [false] 2016-04-04 11:17:29.618898 [DEBUG] sofia.c:4236 ext-rtp-ip [auto-nat] 2016-04-04 11:17:29.618902 [DEBUG] sofia.c:4236 ext-sip-ip [auto-nat] 2016-04-04 11:17:29.618908 [DEBUG] sofia.c:4236 rtp-timeout-sec [300] 2016-04-04 11:17:29.618913 [DEBUG] sofia.c:4236 rtp-hold-timeout-sec [1800] 2016-04-04 11:17:29.618917 [DEBUG] sofia.c:4236 force-register-domain [192.168.199.22] 2016-04-04 11:17:29.618922 [DEBUG] sofia.c:4236 force-subscription-domain [192.168.199.22] 2016-04-04 11:17:29.618927 [DEBUG] sofia.c:4236 force-register-db-domain [192.168.199.22] 2016-04-04 11:17:29.618933 [DEBUG] sofia.c:4236 ws-binding [:5066] 2016-04-04 11:17:29.618939 [DEBUG] sofia.c:4236 wss-binding [:7443] 2016-04-04 11:17:29.618945 [DEBUG] sofia.c:4236 challenge-realm [auto_from] 2016-04-04 11:17:29.618953 [INFO] sofia.c:5513 Setting MAX Auth Validity to 0 Attempts 2016-04-04 11:17:29.619091 [NOTICE] sofia.c:5680 Started Profile internal [sofia_reg_internal]
2016-04-04 11:24:00.891443 [DEBUG] switch_loadable_module.c:735 Chat Thread Started 2016-04-04 11:24:00.891453 [INFO] switch_core.c:2418 FreeSWITCH Version 1.6.5~64bit ( 64bit)
The detailed log is attached in previous emails.
Regards Smile
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, April 08, 2016 11:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log.
On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs@stee.stengg.com <mailto:quanhs@stee.stengg.com>> wrote:
Good. Can you share your scenario ?
Chrome (sipML5) ->FS (1.6.5-64bit Media mode) ->Chrome (sipML5).
Regards Smile
From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Sergey Safarov Sent: Friday, April 08, 2016 3:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
All call with media transcoding enabled. In WebRTC case OPUS <-> G711a
On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs@stee.stengg.com <mailto:quanhs@stee.stengg.com>> wrote: what is setting of inbound-bypass-media and inbound-proxy-media in your case?
Regards Smile
From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@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@stee.stengg.com <mailto: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> [mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto: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 <mailto: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 <mailto:italo@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@stee.stengg.com <mailto: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 <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.
[This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org/>
Hi Michael; It finally functions after adjusting ACL. Great, I almost give up and change to try Verto that FreeSwitch prefers to. Thanks a lot. Next step is to integrate with existing SIP domain or sip phone. Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Tuesday, April 12, 2016 4:00 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP This is never attempting to call the second endpoint: 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:3750 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] It then tries to negotiate the ice session with freeswitch, as it is trying to answer the call locally and send it to voicemail. then we have: 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:3380 sofia/internal/anonymous@192.168.199.22<mailto:sofia/internal/anonymous@192.168.199.22> no suitable candidates found. 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_core_media.c:4473 sofia/internal/anonymous@192.168.199.22<mailto:sofia/internal/anonymous@192.168.199.22> Set 2833 dtmf send payload to 126 recv payload to 126 the issue is your box is on a rfc1918 ip, and the default candidate acl's do not consider those valid. You will need to adjust the acl if you plan on having freeswitch as a server answering webrtc calls from private address space. On Apr 10, 2016, at 10:11 PM, Quan Huo Sheng <quanhs@stee.stengg.com<mailto:quanhs@stee.stengg.com>> wrote: Because existing SIP endpoint doesn’t support SRTP-DTLS, the transcoding (SRTP-DTLS and SRTP-SDES)is required to bridge RTP stream between WebRTC browser and SIP endpoint. 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP: 1709c7c4-fa17-11e5-9184-9d31ebce310e v=0 1709c7c4-fa17-11e5-9184-9d31ebce310e o=- 2269318439683772682 2 IN IP4 127.0.0.1 1709c7c4-fa17-11e5-9184-9d31ebce310e s=- 1709c7c4-fa17-11e5-9184-9d31ebce310e t=0 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=group:BUNDLE audio 1709c7c4-fa17-11e5-9184-9d31ebce310e a=msid-semantic: WMS rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 1709c7c4-fa17-11e5-9184-9d31ebce310e m=audio 1107 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126 1709c7c4-fa17-11e5-9184-9d31ebce310e c=IN IP4 192.168.199.216 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:111 opus/48000/2 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:103 ISAC/16000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:104 ISAC/32000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:9 G722/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:0 PCMU/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:8 PCMA/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:106 CN/32000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:105 CN/16000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:13 CN/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtpmap:126 telephone-event/8000 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp:1110 IN IP4 192.168.199.216 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 1 udp 2122260223 192.168.199.216 1107 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 1 udp 2122194687 10.20.102.216 1108 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 1 udp 2122129151 10.10.13.216 1109 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2985945573 2 udp 2122260222 192.168.199.216 1110 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:2696558014 2 udp 2122194686 10.20.102.216 1111 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1936642402 2 udp 2122129150 10.10.13.216 1112 typ host generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 1 tcp 1518280447 192.168.199.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 1 tcp 1518214911 10.20.102.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 1 tcp 1518149375 10.10.13.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:4286190869 2 tcp 1518280446 192.168.199.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:3996764494 2 tcp 1518214910 10.20.102.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=candidate:1038953874 2 tcp 1518149374 10.10.13.216 0 typ host tcptype active generation 0 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-ufrag:4PWOC8zOrgkF0JcH 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ice-pwd:KoIliPYZgGkk+b6K7bJDPVLK 1709c7c4-fa17-11e5-9184-9d31ebce310e a=fingerprint:sha-256 18:9D:FC:F1:74:38:63:AE:F1:F8:F0:26:F5:83:A8:41:97:53:67:F5:35:FB:7E:F3:06:BF:D5:71:FD:A2:F6:38 1709c7c4-fa17-11e5-9184-9d31ebce310e a=setup:actpass 1709c7c4-fa17-11e5-9184-9d31ebce310e a=mid:audio 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level 1709c7c4-fa17-11e5-9184-9d31ebce310e a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time 1709c7c4-fa17-11e5-9184-9d31ebce310e a=rtcp-mux 1709c7c4-fa17-11e5-9184-9d31ebce310e a=maxptime:60 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 cname:mrCkzHA0WH9Dd+8n 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 msid:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 62a659a4-ef68-469a-871c-0e61ef3dd211 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 mslabel:rYj84RRd3TzyYFyEBpKbVkBuRetSSmtrlY3o 1709c7c4-fa17-11e5-9184-9d31ebce310e a=ssrc:1463663134 label:62a659a4-ef68-469a-871c-0e61ef3dd211 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [NOTICE] switch_channel.c:3501 Hangup sofia/internal/anonymous@192.168.199.22<mailto:sofia/internal/anonymous@192.168.199.22> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 1709c7c4-fa17-11e5-9184-9d31ebce310e 2016-04-04 11:41:28.831214 [DEBUG] switch_ivr_originate.c:1217 sofia/internal/anonymous@192.168.199.22<mailto:sofia/internal/anonymous@192.168.199.22> Media Establishment Failed. ############################################### B leg: 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.471064 [DEBUG] mod_sofia.c:88 sofia/internal/5shm6jpu@tibvb7p7p6od.invalid<mailto:sofia/internal/5shm6jpu@tibvb7p7p6od.invalid> SOFIA INIT 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1228 sip:5shm6jpu@192.168.199.216:1106;transport=wss Setting proxy route to sofia/internal/5shm6jpu@tibvb7p7p6od.invalid<mailto:sofia/internal/5shm6jpu@tibvb7p7p6od.invalid> 1d260ca8-fa17-11e5-91a9-9d31ebce310e 2016-04-04 11:41:28.491143 [DEBUG] sofia_glue.c:1257 sofia/internal/5shm6jpu@tibvb7p7p6od.invalid<mailto:sofia/internal/5shm6jpu@tibvb7p7p6od.invalid> sending invite version: 1.6.5 64bit 1d260ca8-fa17-11e5-91a9-9d31ebce310e Local SDP: 1d260ca8-fa17-11e5-91a9-9d31ebce310e v=0 1d260ca8-fa17-11e5-91a9-9d31ebce310e o=FreeSWITCH 1459715932 1459715933 IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e s=FreeSWITCH 1d260ca8-fa17-11e5-91a9-9d31ebce310e c=IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e t=0 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=msid-semantic: WMS V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp 1d260ca8-fa17-11e5-91a9-9d31ebce310e m=audio 25356 RTP/SAVPF 111 8 102 101 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:111 opus/48000/2 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fmtp:111 minptime=10; useinbandfec=1 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:8 PCMA/8000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:102 telephone-event/48000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtpmap:101 telephone-event/8000 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=fingerprint:sha-256 E7:50:3F:56:DE:21:E5:17:DA:F7:3A:3F:28:F8:BC:2A:C6:E7:E0:67:5C:6C:61:39:00:6D:99:DE:A4:A2:8A:80 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=setup:actpass 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp-mux 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=rtcp:25356 IN IP4 192.168.199.22 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 cname:OFXIzFBV4sOBkyJG 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 msid:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp a0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 mslabel:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzp 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ssrc:2533704824 label:V1fI34R2wy02H1TiwvkVnMYDkt0mDXzpa0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-ufrag:UUr7xgAHzqT3ejiT 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ice-pwd:O6reBS0vbEXl3xRNh8KFfbcE 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 1 udp 659136 192.168.199.22 25356 typ host generation 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=candidate:6905210039 2 udp 659136 192.168.199.22 25356 typ host generation 0 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=ptime:20 1d260ca8-fa17-11e5-91a9-9d31ebce310e a=sendrecv @@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@ Profile: internal 2016-04-04 11:17:29.618618 [DEBUG] sofia.c:4236 debug [9] 2016-04-04 11:17:29.618624 [DEBUG] sofia.c:4236 sip-trace [no] 2016-04-04 11:17:29.618627 [DEBUG] sofia.c:4236 sip-capture [no] 2016-04-04 11:17:29.618630 [DEBUG] sofia.c:4236 watchdog-enabled [no] 2016-04-04 11:17:29.618634 [DEBUG] sofia.c:4236 watchdog-step-timeout [30000] 2016-04-04 11:17:29.618638 [DEBUG] sofia.c:4236 watchdog-event-timeout [30000] 2016-04-04 11:17:29.618642 [DEBUG] sofia.c:4236 log-auth-failures [true] 2016-04-04 11:17:29.618645 [DEBUG] sofia.c:4236 forward-unsolicited-mwi-notify [false] 2016-04-04 11:17:29.618648 [DEBUG] sofia.c:4236 context [public] 2016-04-04 11:17:29.618654 [DEBUG] sofia.c:4236 rfc2833-pt [101] 2016-04-04 11:17:29.618657 [DEBUG] sofia.c:4236 sip-port [5062] 2016-04-04 11:17:29.618661 [DEBUG] sofia.c:4236 dialplan [XML] 2016-04-04 11:17:29.618667 [DEBUG] sofia.c:4236 dtmf-duration [2000] 2016-04-04 11:17:29.618677 [DEBUG] sofia.c:4236 inbound-codec-prefs [OPUS] 2016-04-04 11:17:29.618684 [DEBUG] sofia.c:4236 outbound-codec-prefs [OPUS] 2016-04-04 11:17:29.618690 [DEBUG] sofia.c:4236 rtp-timer-name [soft] 2016-04-04 11:17:29.618694 [DEBUG] sofia.c:4236 rtp-ip [192.168.199.22] 2016-04-04 11:17:29.618698 [DEBUG] sofia.c:4236 sip-ip [192.168.199.22] 2016-04-04 11:17:29.618702 [DEBUG] sofia.c:4236 hold-music [local_stream://moh] 2016-04-04 11:17:29.618720 [DEBUG] sofia.c:4236 apply-nat-acl [nat.auto] 2016-04-04 11:17:29.618734 [ERR] sofia.c:5185 Not adding acl nat.auto because it's the local network 2016-04-04 11:17:29.618739 [DEBUG] sofia.c:4236 apply-inbound-acl [domains] 2016-04-04 11:17:29.618745 [DEBUG] sofia.c:4236 local-network-acl [localnet.auto] 2016-04-04 11:17:29.618749 [DEBUG] sofia.c:4236 record-path [/usr/local/freeswitch/recordings] 2016-04-04 11:17:29.618754 [DEBUG] sofia.c:4236 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2016-04-04 11:17:29.618763 [DEBUG] sofia.c:4236 manage-presence [true] 2016-04-04 11:17:29.618768 [DEBUG] sofia.c:4236 presence-hosts [192.168.199.22,192.168.199.22] 2016-04-04 11:17:29.618774 [DEBUG] sofia.c:4236 presence-privacy [false] 2016-04-04 11:17:29.618778 [DEBUG] sofia.c:4236 inbound-codec-negotiation [generous] 2016-04-04 11:17:29.618783 [DEBUG] sofia.c:4236 tls [true] 2016-04-04 11:17:29.618791 [DEBUG] sofia.c:4236 tls-only [false] 2016-04-04 11:17:29.618798 [DEBUG] sofia.c:4236 tls-bind-params [transport=tls] 2016-04-04 11:17:29.618804 [DEBUG] sofia.c:4236 tls-sip-port [5063] 2016-04-04 11:17:29.618810 [DEBUG] sofia.c:4236 tls-passphrase [] 2016-04-04 11:17:29.618816 [DEBUG] sofia.c:4236 tls-verify-date [true] 2016-04-04 11:17:29.618822 [DEBUG] sofia.c:4236 tls-verify-policy [none] 2016-04-04 11:17:29.618829 [ERR] sofia_glue.c:329 Invalid tls-verify-policy value: none 2016-04-04 11:17:29.618835 [DEBUG] sofia.c:4236 tls-verify-depth [2] 2016-04-04 11:17:29.618844 [DEBUG] sofia.c:4236 tls-verify-in-subjects [] 2016-04-04 11:17:29.618850 [DEBUG] sofia.c:4236 tls-version [tlsv1,tlsv1.1,tlsv1.2] 2016-04-04 11:17:29.618860 [DEBUG] sofia.c:4236 tls-ciphers [ALL:!ADH:!LOW:!EXP:!MD5:@STRENGTH] 2016-04-04 11:17:29.618868 [DEBUG] sofia.c:4236 inbound-late-negotiation [true] 2016-04-04 11:17:29.618872 [DEBUG] sofia.c:4236 inbound-zrtp-passthru [true] 2016-04-04 11:17:29.618876 [DEBUG] sofia.c:4236 nonce-ttl [60] 2016-04-04 11:17:29.618881 [DEBUG] sofia.c:4236 auth-calls [true] 2016-04-04 11:17:29.618887 [DEBUG] sofia.c:4236 inbound-reg-force-matching-username [true] 2016-04-04 11:17:29.618893 [DEBUG] sofia.c:4236 auth-all-packets [false] 2016-04-04 11:17:29.618898 [DEBUG] sofia.c:4236 ext-rtp-ip [auto-nat] 2016-04-04 11:17:29.618902 [DEBUG] sofia.c:4236 ext-sip-ip [auto-nat] 2016-04-04 11:17:29.618908 [DEBUG] sofia.c:4236 rtp-timeout-sec [300] 2016-04-04 11:17:29.618913 [DEBUG] sofia.c:4236 rtp-hold-timeout-sec [1800] 2016-04-04 11:17:29.618917 [DEBUG] sofia.c:4236 force-register-domain [192.168.199.22] 2016-04-04 11:17:29.618922 [DEBUG] sofia.c:4236 force-subscription-domain [192.168.199.22] 2016-04-04 11:17:29.618927 [DEBUG] sofia.c:4236 force-register-db-domain [192.168.199.22] 2016-04-04 11:17:29.618933 [DEBUG] sofia.c:4236 ws-binding [:5066] 2016-04-04 11:17:29.618939 [DEBUG] sofia.c:4236 wss-binding [:7443] 2016-04-04 11:17:29.618945 [DEBUG] sofia.c:4236 challenge-realm [auto_from] 2016-04-04 11:17:29.618953 [INFO] sofia.c:5513 Setting MAX Auth Validity to 0 Attempts 2016-04-04 11:17:29.619091 [NOTICE] sofia.c:5680 Started Profile internal [sofia_reg_internal] 2016-04-04 11:24:00.891443 [DEBUG] switch_loadable_module.c:735 Chat Thread Started 2016-04-04 11:24:00.891453 [INFO] switch_core.c:2418 FreeSWITCH Version 1.6.5~64bit ( 64bit) The detailed log is attached in previous emails. Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Friday, April 08, 2016 11:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP Seriously, use verto... it will be much cleaner... also, the issue you are likely having has to do with codec negotiation settings, but we can't say for sure without seeing a debug log. On Apr 8, 2016, at 4:16 AM, Quan Huo Sheng <quanhs@stee.stengg.com<mailto:quanhs@stee.stengg.com>> wrote: Good. Can you share your scenario ? Chrome (sipML5) ->FS (1.6.5-64bit Media mode) ->Chrome (sipML5). Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Friday, April 08, 2016 3:17 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP All call with media transcoding enabled. In WebRTC case OPUS <-> G711a On Fri, Apr 8, 2016, 09:34 Quan Huo Sheng <quanhs@stee.stengg.com<mailto:quanhs@stee.stengg.com>> wrote: what is setting of inbound-bypass-media and inbound-proxy-media in your case? Regards Smile From: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@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@stee.stengg.com<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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> [mailto: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> [mailto: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<mailto: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> [mailto: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<mailto: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<mailto:italo@freeswitch.org> IRC chat.freenode.net<http://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@stee.stengg.com<mailto: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<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. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group--- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://confluence.freeswitch.org<http://confluence.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org<http://www.freeswitch.org/>
participants (5)
-
Michael Jerris
-
Quan Huo Sheng
-
saurabh verrma
-
Sergey Safarov
-
Ítalo Rossi