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No audio from webrtc2sip to Chrome when attaching a legacy device #66

Open GoogleCodeExporter opened 8 years ago

GoogleCodeExporter commented 8 years ago 
What steps will reproduce the problem?
1.We dial from Chrome 25 using SIPML5 to a legacy device.
2.Video is disabled and Breaker enabled. No media coder on webrtc2sip. Audio is 
G.711.
3. The legacy device does not use ICE nor SRTP. RTCP-MUX is not supported.

What is the expected output? What do you see instead?
Audio goes from Chrome to webrtc2sip and from there to the legacy device. Audio 
also goes from the legacy device to webrtc2sip. No audio is sent from 
webrtc2sip to Chrome.

What version of the product are you using? On what operating system?
Chrome version is 25. SIPML5 is run from the SIPML5.og. webrtc2Sip version is 
2.0 - a week old.

Please provide any additional information below.
Similar configuration where the legacy device is Boghe works well.

Original issue reported on code.google.com by avishay....@gmail.com on 5 Mar 2013 at 4:36

Attachments:

GoogleCodeExporter commented 8 years ago 
what's your doubango version?

Original comment by boss...@yahoo.fr on 5 Mar 2013 at 5:26

GoogleCodeExporter commented 8 years ago 
My doubango version number is: *2.0.0.814*

*Thanks*

*Avishay*

Original comment by avishay....@gmail.com on 6 Mar 2013 at 10:27

GoogleCodeExporter commented 8 years ago 
the question about Doubango was for the SVN revision. If you don't have the 
latest revisionn you should update your code

Original comment by boss...@yahoo.fr on 6 Mar 2013 at 5:42

GoogleCodeExporter commented 8 years ago 
I ran 'svn info' in the directory where I downloaded doubango and this is
what I got:

Path: .

URL: http://doubango.googlecode.com/svn/branches/2.0/doubango

Repository Root: http://doubango.googlecode.com/svn

Repository UUID: c7b0ae48-b0eb-11de-8bdf-3374eb5c7316

Revision: 814

Node Kind: directory

Schedule: normal

Last Changed Author: bossiel@yahoo.fr

Last Changed Rev: 814

Last Changed Date: 2013-02-06 13:12:25 +0200 (Wed, 06 Feb 2013)

The latest version in the svn is 832 now. So, I understand that you suggest
that I will update the code. right?

Original comment by avishay....@gmail.com on 6 Mar 2013 at 7:28

GoogleCodeExporter commented 8 years ago 
yes

Original comment by boss...@yahoo.fr on 6 Mar 2013 at 9:19

GoogleCodeExporter commented 8 years ago 
We upgraded Doubango and WebRtc2SIP to the latest version and it did no help. 
Then, we found that the source of problem is that our device sent an initial 
UDP packet to the WebRtc2Sip (Breaker)that made it stopped receiving packets 
from our device.

That packet is an empty UDP packet, sent as a NAT keepalive packet, as 
specified in http://www.rfc-editor.org/rfc/rfc6263.txt

"4.1.  Empty (0-Byte) Transport Packet

   The application sends an empty transport packet (e.g., UDP packet,
   Datagram Congestion Control Protocol (DCCP) packet)."

In this case the keepalive was not really necessary. Yet, we did not expect it 
to make the WebRtc2Sip stop receiving packets from our device.

When I configured our device not to send those keepalive packets, audio packets 
went in all 4 legs.

Thanks
Avishay

Original comment by avishay....@gmail.com on 11 Mar 2013 at 6:17

GoogleCodeExporter commented 8 years ago 
Good catch. Try to remove condition at 
https://code.google.com/p/doubango/source/browse/branches/2.0/doubango/tinyNET/s
rc/tnet_transport_poll.c?r=834#720 and rebuild the source

Original comment by boss...@yahoo.fr on 12 Mar 2013 at 12:17