(asterisk + sipml5 + webrtc2sip) audio and video not streaming through connected call

[Jason]

Sorry if this is a duplicate thread, I don't think my topic was actually posted.

I have a video call being connected, though taking 20-30 seconds, it connects. But then, the audio and video streams don't seem to be going through.

I have a virtual machine running debian, I followed this tutorial to setup asterisk and webrtc2sip: http://linux.autostatic.com/asterisk-and-sipml5-interoperability

each machine is on a business wifi network with IP addresses 10.1.7.*, so I was trying to disable STUN. when changing config.xml <enable-icestun>no</enable-icestun>

this reduced the connection time from 45 seconds to a minute to what it is now. I might have to setup my own STUN server on the same VM?

asterisk sip.conf

http://pastebin.com/kiYbbBrm

asterisk extensions.conf

http://pastebin.com/fdnxB37m

webrtc2sip config.xml

http://pastebin.com/MdKDZdX2

webrtc2sip screen session 

http://pastebin.com/CiBp5L6G

asterisk screen session  (there isn't a /var/log/asterisk/full file, is there another location I can look?)

http://pastebin.com/UgLjuSJC

chrome canary console log 1

http://pastebin.com/wj8j2LSF

chrome canary console log 2

http://pastebin.com/7YZPTKLB

sipml5 web config  (http://sipml5.org/call.htm?svn=179) I will put this on the virtual machine in the end, but for testing I thought this was ok.

http://imageshack.us/f/690/wiu.png/

If I am not supplying enough information, then let me know, and I will gladly share more.

[Mamadou DIOP]

First message in the group is moderated and have to be manually validated.

We'll need full logs for webrtc2sip. Just redirect the console logs to a file by adding at the end of the command used to start the gateway "> logs.txt"

You said "this reduced the connection time from 45 seconds to a minute to what it is now". ???

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[Jason]

Sorry for the misunderstanding, I was referring to https://groups.google.com/forum/#!topic/discuss-webrtc/oj6CR1ftnWc. (It took 45 seconds to a minute for a call to go through)

I then changed <enable-icestun>yes</enable-icestun>  to  <enable-icestun>no</enable-icestun>

so, after the change and restarting webrtc2sip, now when one browser calls the other, the time between clicking the call button and when the other rings, answers and then a call is established, is between 20-30 seconds.

I realize that this should be taking even less time than that.

when not using a screen session and simply running webrtc2sip --config=/usr/local/etc/webrtc2sip/config.xml > webrtc2sip.txt

*******************************************************************

Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>

PRODUCT: webrtc2sip

HOME PAGE: http://webrtc2sip.org

LICENCE: GPLv3 or proprietary

VERSION: 2.5.1

'quit' to quit the application.

*******************************************************************

SSL is enabled :)

DTLS supported: yes

DTLS-SRTP supported: yes

***ERROR: function: "tsk_params_get_param_value()"

file: "src/tsk_params.c"

line: "219"

MSG: Invalid parameter

***ERROR: function: "tsk_params_get_param_value()"

file: "src/tsk_params.c"

line: "219"

MSG: Invalid parameter

warning: The VAD has been replaced by a hack pending a complete rewrite

warning: The VAD has been replaced by a hack pending a complete rewrite

warning: The VAD has been replaced by a hack pending a complete rewrite

warning: The VAD has been replaced by a hack pending a complete rewrite

warning: The VAD has been replaced by a hack pending a complete rewrite

***ERROR: function: "_trtp_manager_recv_data()"

file: "src/trtp_manager.c"

line: "420"

MSG: srtp_unprotect(RTP) failed with error code=9, seq_num=2739

***ERROR: function: "_trtp_manager_recv_data()"

file: "src/trtp_manager.c"

line: "420"

MSG: srtp_unprotect(RTP) failed with error code=9, seq_num=2798

***ERROR: function: "tsdp_header_M_get_holdresume_att()"

file: "src/headers/tsdp_header_M.c"

line: "759"

MSG: Invalid parameter

***ERROR: function: "_trtp_manager_ice_init()"

file: "src/trtp_manager.c"

line: "675"

MSG: ICE context not ready

***ERROR: function: "trtp_manager_start()"

file: "src/trtp_manager.c"

line: "1205"

MSG: _trtp_manager_ice_init() failed

warning: The VAD has been replaced by a hack pending a complete rewrite

[Mamadou DIOP]

edit config.xml and change debug level to INFO.

On Jul 11, 2013, at 5:58 PM, Jason <eleva...@gmail.com> wrote:

Sorry for the misunderstanding, I was referring to https://groups.google.com/forum/#!topic/discuss-webrtc/oj6CR1ftnWc. (It took 45 seconds to a minute for a call to go through)

I then changed <enable-icestun>yes</enable-icestun>  to  <enable-icestun>no</enable-icestun>

so, after the change and restarting webrtc2sip, now when one browser calls the other, the time between clicking the call button and when the other rings, answers and then a call is established, is between 20-30 seconds.

I realize that this should be taking even less time than that.

when not using a screen session and simply running webrtc2sip --config=/usr/local/etc/webrtc2sip/config.xml > webrtc2sip.txt

It seems that the only thing that shows up in the log.txt is after I quit is:

*******************************************************************

Copyright (C) 2012-2013 Doubango Telecom <http://www.doubango.org>

PRODUCT: webrtc2sip

HOME PAGE: http://webrtc2sip.org

LICENCE: GPLv3 or proprietary

VERSION: 2.5.1

'quit' to quit the application.

*******************************************************************

SSL is enabled :)

DTLS supported: yes

DTLS-SRTP supported: yes

On Thursday, July 11, 2013 10:10:08 AM UTC-5, Mamadou wrote:

[Mamadou]

The same STUN/TURN timeout happens at the caller and callee sides because you don't have access to internet or default servers are down (few chances).
There is no API function to disable STUN/TURN on chrome. You can:
- Edit the code to use null iceServers: https://code.google.com/p/sipml5/source/browse/trunk/src/tinyMEDIA/src/tmedia_session_jsep.js?r=195#657
- Or, setup your own local STUN/TURN server and set it at http://sipml5.org/debug/expert.htm
webrtc2sip must not take more than few milliseconds to forward a call.

--
Mamadou DIOP - Technology Evangelist
Doubango Telecom - Paris, France
http://www.doubango.org
Click here to call me!

[Jason]

Sorry about that, I had to figure out screen session logging, but I know now.

here is the INFO level debug info.

http://pastebin.com/VPFmG02K

[Jason]

Thanks, I will try to set up the code on the VM webserver with null iceServers

[Jason]

I put the sipml5 code on the VM webserver, and changed line 657 from tmedia_session_jsep.js from

                { iceServers: o_iceServers },

to 

                null,

The connection time is now virtually instant, though I am still not getting video to or from the other end.

Here is the webrtc2sip log for the new configuration:

INFO level debug

http://pastebin.com/dwfgzzg6

ERROR level debug (the INFO level debug seemed quite large)

http://pastebin.com/nfH1wV7R

[Jason]

When trying to use a local stun server instead, with line 657 now changed to ->     { iceServers: [{ url: 'stun:10.1.7.55:3478'}] },

this open source stun server doesn't have TLS implementation, and is only STUN server, it is not a TURN server

I'm not sure if that matters, but I wanted to make sure.

I get this ERROR level debug appended to what I posted in the last post

[Jason]

Just in case, I changed to trying https://code.google.com/p/rfc5766-turn-server/

Just because it is STUN and TURN. It didn't fix the streaming problem. The calls still connect, but the audio and video doesn't get to the other side.

I should just be able to have STUN off and have it all working, since each browser is on the same network as the asterisk/webrtc2sip virtual machine.

Is there any other reasons for video not streaming that anyone knows of?