search

IliyaPIS / sipml5

Automatically exported from code.google.com/p/sipml5
BSD 3-Clause "New" or "Revised" License
0 stars 0 forks source link

One way video, any browser, webrtc2sip and Asterisk to softphone #228

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago 
The problem occurs when I make or receive a call from Firefox or Chrome, via 
webrtc2sip and asterisk to a softphone (Ekiga). I've attached the sipml5 log 
file.

I think I'm getting audio in both directions (the mic is faulty on one 
machine), but video only in one direction. Video is sent from the web browser 
to the softphone, but doesn't get back to the web browser. There are messages 
appearing in the webrtc2sip debug:

*[DOUBANGO INFO]: Incoming SIP INFO(picture_fast_update)
*[DOUBANGO INFO]: ***IDR request tooo close(2 ms)...ignoring****

and 

**[DOUBANGO WARN]: function: "tdav_codec_h263_decode()"
file: "src/codecs/h263/tdav_codec_h263.c"
line: "440"
MSG: Buffer overflow

Also, before video starts I get thousands of 

*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"
*[DOUBANGO INFO]: Sending DNS query to "192.168.1.250"

appearing in the webrtc2sip log.

The asterisk config allows g722 and h263 only.

It's probably something that isn't configured correctly, or a mistake in my 
sipml5 code, but I'd be grateful for some help solving this problem.

thanks,

Charles.

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?

What version of the product are you using? On what operating system?

Please provide any additional information below.

Original issue reported on code.google.com by charles....@gmail.com on 15 Jun 2015 at 3:53

Attachments:

GoogleCodeExporter commented 8 years ago 
oops. I didn't fill out all the requested information.

webrtc2sip v2.6.0
sipml5 - trunk, downloaded 11/6/15
operating system - sipml5 on Windows 7, Firefox 38.0.5. Webrtc2sip is on Ubuntu 
Trusty
asterisk is v13.3.2

Original comment by charles....@gmail.com on 16 Jun 2015 at 8:53

GoogleCodeExporter commented 8 years ago 
I've tried using h264 instead of h263 and get the same problem:

SEND: SIP/2.0 200 Ok
Via: SIP/2.0/UDP 
192.168.1.241:5060;rport=5060;received=192.168.1.241;branch=z9hG4bK323ae356
From: <sip:60007@192.168.1.241>;tag=as76e4f0da
To: <sip:60006@192.168.1.241>;tag=2292713159
Contact: <sip:60006@192.168.1.241:10060;transport=udp>
Call-ID: 5202c1ee-4142-be48-1504-9ebe47099bab
CSeq: 365 INFO
Content-Length: 0

*[DOUBANGO INFO]: Incoming SIP INFO(picture_fast_update)
*[DOUBANGO INFO]: ***IDR request tooo close(1 ms)...ignoring****
***[DOUBANGO ERROR]: function: "tdav_codec_h264_decode()"
file: "src/codecs/h264/tdav_codec_h264.c"
line: "384"
MSG: BufferOverflow

Original comment by charles....@gmail.com on 18 Jun 2015 at 4:13