Re: No video output Webrtc2sip + Asterisk

[Hadi Ams]

Hi,

When you are using webrtc2sip , you don't have to patch Asterisk for VP8 .

VP8 support in Asterisk is experimental and does not work that well .

Instead you should enable the media coder in webrtc2sip and use other video codecs that Asterisk naturally supports like H.264 .

So offered video codec from chrome will be VP8 and on Asterisk side it will be H.264 and webrtc2sip will take care of transcoding ;)

Good luck

[AutoStatic]

Would you mind sharing your patch? And try directmedia = outgoing for the peer Chrome uses. Asterisk rewrites its invites but that's not necessary if webrt2sip is on the same network (so no NAT).
VP8 support in Asterisk can be useful if SIP clients that connect to Asterisk only support VP8 (like the current Linphone mobile app).

Jeremy

[navaismo]

Hi  thanks for the reply, the mediacoder is enabled and for the debug I think the gateway try to use h264 but fails and there is no video.

[navaismo]

Thanks for the reply, I'll try that, however there is no Nat involved. About the patch I have downloaded from your page(I guess this is your blog page http://linux.autostatic.com/asterisk-and-sipml5-interoperability) and it works with the latest asterisk at least to define allow=vp8 as video codec.

 

[AutoStatic]

On Saturday, 19 January 2013 15:47:14 UTC+1, navaismo wrote:

El sábado, 19 de enero de 2013 01:29:44 UTC-6, AutoStatic escribió:

Would you mind sharing your patch? And try directmedia = outgoing for the peer Chrome uses. Asterisk rewrites its invites but that's not necessary if webrt2sip is on the same network (so no NAT).
VP8 support in Asterisk can be useful if SIP clients that connect to Asterisk only support VP8 (like the current Linphone mobile app).

Jeremy

 

 

Thanks for the reply, I'll try that, however there is no Nat involved.

So the two endpoints (sipml5 and the other SIP endpoint you're using), webrtc2sip and Asterisk are all on the same network?

About the patch I have downloaded from your page(I guess this is your blog page http://linux.autostatic.com/asterisk-and-sipml5-interoperability) and it works with the latest asterisk at least to define allow=vp8 as video codec.

Yes that's my blog. Good to know that it also works with 11.2.0.

Jeremy

 

[navaismo]

El sábado, 19 de enero de 2013 08:51:37 UTC-6, AutoStatic escribió:

On Saturday, 19 January 2013 15:47:14 UTC+1, navaismo wrote:

El sábado, 19 de enero de 2013 01:29:44 UTC-6, AutoStatic escribió:

Would you mind sharing your patch? And try directmedia = outgoing for the peer Chrome uses. Asterisk rewrites its invites but that's not necessary if webrt2sip is on the same network (so no NAT).
VP8 support in Asterisk can be useful if SIP clients that connect to Asterisk only support VP8 (like the current Linphone mobile app).

Jeremy

 

 

Thanks for the reply, I'll try that, however there is no Nat involved.

So the two endpoints (sipml5 and the other SIP endpoint you're using), webrtc2sip and Asterisk are all on the same network?

Yes all peers are in the same network. I saw this error in the log:

[h264 @ 0x7f5cfc052f40] Warning: not compiled with thread support, using thread emulation
***ERROR: function: "tdav_codec_h264_sdp_att_match()"
file: "src/codecs/h264/tdav_codec_h264.c"
line: "415"
MSG: Not valid profile-level: redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
*INFO: Trying to match [fmtp:redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0]
***ERROR: function: "tdav_codec_h264_sdp_att_match()"
file: "src/codecs/h264/tdav_codec_h264.c"
line: "415"

 

Maybe that's why I cant send/see the video, but how I can fix it?

[Mamadou DIOP]

The error seems to be clear: the fmt from your SIP-legacy network is not correct.

"profile-level-id" is required but missing.

--
 
 

[navaismo]

El sábado, 19 de enero de 2013 14:16:00 UTC-6, Mamadou escribió:

The error seems to be clear: the fmt from your SIP-legacy network is not correct.

"profile-level-id" is required but missing.

 

 

Hi, I don't know what that means and if is related to the video transcoding error. Since audio is working great. Can you elaborate more please?

 

Thanks :D

[navaismo]

Hi,

Still no luck i have configured all based on the blog of Autostatic,  and again the video is failing to show. Only the green square and the h264 errors on the media gateway.

All the test are in the same machine, I mean, the Asterisk, the media gateway, the sipml5 demo and the generic sip extension are in the same machine using same IP differents ports. Is this the Issue? About the profile-level-id When i enable the sip debug in asterisk I can see it in the sdp so dont know more than that.

If you can share some hints I can search how to solve, since I'm new to the media gateway usage.

Best Regards!

[navaismo]

Another Update: I have tested the self video with this URL http://blog.tmcnet.com/blog/tom-keating/webrtctest.html and the video still green. 

Since that URL was tested and seems to be working for another people and most important that demo is running out of my scope and I'm just a client  I think the issue came from my WebCam, I mean is unsupported.

[AutoStatic]

On Monday, 21 January 2013 23:12:33 UTC+1, navaismo wrote:

Hi,

Still no luck i have configured all based on the blog of Autostatic,  and again the video is failing to show. Only the green square and the h264 errors on the media gateway.

Where do you get the green square? What kind of legacy SIP clients are you using?
 

All the test are in the same machine, I mean, the Asterisk, the media gateway, the sipml5 demo and the generic sip extension are in the same machine using same IP differents ports. Is this the Issue?

Probably not as Mamadou already explained. And what kind of "generic sip extension" is on the same machine?

About the profile-level-id When i enable the sip debug in asterisk I can see it in the sdp so dont know more than that.

https://supportforums.cisco.com/community/netpro/collaboration-voice-video/telepresence/blog/2011/01/14/video--telepresence-sip-h264-profile-level-id

Apparently your legacy SIP client needs to include a profile-level-id and it doesn't while webrtc2sip expects one. So then webrtc2sip spits out errors and refuses to transport the video. At least that's what I think is happening, but I could be wrong, videocalling and WebRTC are relatively new things for me.


Jeremy

[navaismo]

Where do you get the green square? What kind of legacy SIP clients are you using?

I get the green square when I have tried with video enable, The green square is supposed  to be mi video output. As I said before also trying with this link http://blog.tmcnet.com/blog/tom-keating/webrtctest.html  My video Output is a big green square. 

And I'm using SIP Softphones registered to Asterisk.

 

[AutoStatic]

On Wednesday, 23 January 2013 18:22:09 UTC+1, navaismo wrote:

Where do you get the green square? What kind of legacy SIP clients are you using?

I get the green square when I have tried with video enable, The green square is supposed  to be mi video output. As I said before also trying with this link http://blog.tmcnet.com/blog/tom-keating/webrtctest.html  My video Output is a big green square. 

Ah now I understand :) All is on the same machine which is presumably an Ubuntu 12.04 install.
sudo usermod -a -G video yourusername
should do the trick. If that doesn't work your webcam is probably unsupported.
 

And I'm using SIP Softphones registered to Asterisk.

What kind of SIP softphones? Xlite, Boghe, Linphone, Ekiga, SFLPhone, Empathy, Jitsi, Zoiper, Phoner, Twinkle, ...?

Jeremy

 

[navaismo]

El miércoles, 23 de enero de 2013 14:20:20 UTC-6, AutoStatic escribió:

On Wednesday, 23 January 2013 18:22:09 UTC+1, navaismo wrote:

Where do you get the green square? What kind of legacy SIP clients are you using?

I get the green square when I have tried with video enable, The green square is supposed  to be mi video output. As I said before also trying with this link http://blog.tmcnet.com/blog/tom-keating/webrtctest.html  My video Output is a big green square. 

Ah now I understand :) All is on the same machine which is presumably an Ubuntu 12.04 install.
sudo usermod -a -G video yourusername
should do the trick. If that doesn't work your webcam is probably unsupported.
 

Nope, my OS it's Fedora 17 and my user can use the video for others applications, also tested under root. I believe that my 2cents webcam is unsupported too.

 

And I'm using SIP Softphones registered to Asterisk.

What kind of SIP softphones? Xlite, Boghe, Linphone, Ekiga, SFLPhone, Empathy, Jitsi, Zoiper, Phoner, Twinkle, ...?

Tested with: Linphone, SFLphone & Jitsi

Thanks again.

 

[Deekshitha Sundaran]

Hi,

            Is video conferencing supported in telepresence.If so , what should I enable to make video conferencing work.I have tested audio conferencing it works fine. I would like to know about videoconferencing in telepresence.