[Freeswitch-users] Webrtc no video

Greg greg at jazzmessengers.com
Wed Jan 7 11:23:40 MSK 2015


Thank you for your answer !

I will try to talk with the FS consulting service.

El 07/01/2015 a las 3:25, Anthony Minessale escribió:
> The video support in FS is undergoing massive modification.
> Support is currently limited to same codecs and in limited setups.
>
> When 1.6 beta emerges, give it a try.
>
> On Tue, Jan 6, 2015 at 6:23 PM, <greg at jazzmessengers.com 
> <mailto:greg at jazzmessengers.com>> wrote:
>
>     Hi folks,
>
>     I have just installed freeswitch with a basic configuration. Actually,
>     i trying to get to work a call with video. I'm using sipjs to set up
>     my client. Each call has audio but no video.
>
>     After some days checking the docs and some test, i don't really know
>     where could come from this problem.
>     Maybe the codec.
>
>     Here, from fs_cli, the codecs list i have installed:
>
>     ===
>     codec,ADPCM (IMA),mod_spandsp
>     codec,AMR,mod_amr
>     codec,G.711 alaw,CORE_PCM_MODULE
>     codec,G.711 ulaw,CORE_PCM_MODULE
>     codec,G.722,mod_spandsp
>     codec,G.723.1 6.3k,mod_g723_1
>     codec,G.726 16k,mod_spandsp
>     codec,G.726 16k (AAL2),mod_spandsp
>     codec,G.726 24k,mod_spandsp
>     codec,G.726 24k (AAL2),mod_spandsp
>     codec,G.726 32k,mod_spandsp
>     codec,G.726 32k (AAL2),mod_spandsp
>     codec,G.726 40k,mod_spandsp
>     codec,G.726 40k (AAL2),mod_spandsp
>     codec,G.729,mod_g729
>     codec,GSM,mod_spandsp
>     codec,H.261 Video (passthru),mod_h26x
>     codec,H.263 Video (passthru),mod_h26x
>     codec,H.263+ Video (passthru),mod_h26x
>     codec,H.263++ Video (passthru),mod_h26x
>     codec,H.264 Video (passthru),mod_h26x
>     codec,LPC-10,mod_spandsp
>     codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>     codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
>     codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
>     codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
>     codec,Speex,CORE_SPEEX_MODULE
>     27 total.
>     ====
>
>
>     I tried to call to soft sip (x-lite on my computer, zoiper on my
>     iphone 4s).
>     here some log
>
>     call accepted:
>
>     ==============
>     2015-01-07 01:14:43.534992 [NOTICE] sofia.c:7475 Channel
>     [sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178>] has been answered
>     2015-01-07 01:14:43.534992 [DEBUG] switch_channel.c:3689
>     (sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178>) Callstate Change RINGING
>     -> ACTIVE
>     2015-01-07 01:14:43.554994 [DEBUG] switch_core_codec.c:246
>     sofia/internal/1000 at 37.187.113.94 <mailto:1000 at 37.187.113.94>
>     Restore previous codec PCMA:8.
>     2015-01-07 01:14:43.554994 [DEBUG] mod_sofia.c:780 Local SDP
>     sofia/internal/1000 at 37.187.113.94 <mailto:1000 at 37.187.113.94>:
>     v=0
>     o=FreeSWITCH 1420564044 1420564046 IN IP4 37.187.113.94
>     s=FreeSWITCH
>     c=IN IP4 37.187.113.94
>     t=0 0
>     m=audio 25628 RTP/AVP 8 101
>     a=rtpmap:8 PCMA/8000
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-16
>     a=ptime:20
>     a=sendrecv
>     ===============
>
>     when i active the video from one of the client:
>
>
>     ====
>
>     2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send
>     signal sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> [BREAK]
>     2015-01-07 01:18:23.094992 [DEBUG] switch_core_session.c:1053 Send
>     signal sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> [BREAK]
>     2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6614 Channel
>     sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> entering state
>     [received][100]
>     2015-01-07 01:18:23.114995 [DEBUG] sofia.c:6624 Remote SDP:
>     v=0
>     o=- 13065063286487275 4 IN IP4 62.57.238.211
>     s=X-Lite release 4.7.1 stamp 74247
>     c=IN IP4 62.57.238.211
>     t=0 0
>     m=audio 55746 RTP/AVP 8 0 101 125 100 9
>     a=rtpmap:101 telephone-event/8000
>     a=fmtp:101 0-15
>     a=rtpmap:125 opus/48000/2
>     a=fmtp:125 useinbandfec=1
>     a=rtpmap:100 speex/16000
>     m=video 58218 RTP/AVP 115 34
>     a=rtpmap:115 H263-1998/90000
>     a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1
>     a=rtpmap:34 H263/90000
>     a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2
>     a=rtcp-fb:* nack pli
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3682 Audio
>     Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3543 Set
>     telephone-event payload to 101
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [opus:125:48000:20:0:2]/[PCMA:8:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [opus:125:48000:20:0:2]/[PCMU:0:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [speex:100:16000:20:0:1]/[PCMA:8:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [speex:100:16000:20:0:1]/[PCMU:0:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3627 Audio
>     Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:3881 Set 2833
>     dtmf send payload to 101
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_media.c:5124 Audio
>     params are unchanged for
>     sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178>.
>     2015-01-07 01:18:23.114995 [DEBUG] sofia.c:7259 Processing updated SDP
>     2015-01-07 01:18:23.114995 [DEBUG] switch_core_session.c:1053 Send
>     signal sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> [BREAK]
>     2015-01-07 01:18:23.134997 [DEBUG] sofia.c:6614 Channel
>     sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> entering state
>     [completed][200]
>     2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send
>     signal sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> [BREAK]
>     2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send
>     signal sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> [BREAK]
>     2015-01-07 01:18:23.314994 [DEBUG] switch_core_session.c:1053 Send
>     signal sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> [BREAK]
>     2015-01-07 01:18:23.334992 [DEBUG] sofia.c:6614 Channel
>     sofia/internal/sip:1002 at 62.57.238.211:36178
>     <http://sip:1002@62.57.238.211:36178> entering state [ready][200]
>
>     ====
>
>
>     Thank in advance for your help
>
>     Greg
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at 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 at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
>http://freeswitch.org/http://cluecon.com/> http://twitter.com/FreeSWITCH
> ☞ irc.freenode.net <http://irc.freenode.net> #freeswitch ☞ 
> _http://freeswitch.org/g+_
>
> ClueCon Weekly Development Call
> ☎ sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>  ☎ +19193869900
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at 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 at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20150107/18bf065c/attachment.html 


Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list