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---