No VT call possible outside NAT.

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What steps will reproduce the problem?
1.  Changed talk_base::SocketAddress stun_addr("stun.l.google.com",19302); to  
talk_base::SocketAddress stun_addr("relay.google.com ", 19302);
2. Made a vcall(from comand prompt) to my friend outside my nat. 
3. My friend is logged in gmail on browser(supporting voice/video).

What is the expected output? What do you see instead?
He was able to reveive the call but couldn't see the audio/video(rtp dump files 
in sample application).

What version of the product are you using? On what operating system?
libjingle 0.5.1

Please provide any additional information below.
Stun it may not work as NAT will block the udp packets but Turn should work. 

Is Turn server not implemented in libjingle 0.5.1??

Original issue reported on code.google.com by satya.bh...@gmail.com on 3 Dec 2010 at 11:07

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[deleted comment]

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sorry i mentioned **reveive** the call...but its He was able to **receive** the 
call but couldn't see the audio/video(rtp dump files in sample application). 

Original comment by satya.bh...@gmail.com on 3 Dec 2010 at 11:09

[GoogleCodeExporter]

Just wanted to clarify, were you using the "call" program? if so, this is 
expected behavior, because libjingle code base doesn't have a voice/video media 
engine. It only has the file engine. So you need to feed in a media file when 
using it.
Like:
call --videoinput=..\..\..\session\phone\testdata\video.rtpdump 
--voiceinput=..\..\..\session\phone\testdata\voice.rtpdump

Original comment by jun...@google.com on 10 Dec 2010 at 9:30

• Changed state: Done

[GoogleCodeExporter]

Before putting up this question i made relevant changes in the call_main.cc 
file giving audio/video RTPdump files as input. But my problem is it is not 
bypassing the NAT. So my friend outside my NAT is not able to see 
audio/video(dump files).
I have 2 questions related to it.

1. Is your TURN code working in libjingle? I am not sure but i don't find the 
code for turn compliant to RFC 5766.

2. If TURN code is working then HTTP tunneling related code in the libjingle is 
not working. What changes need to be made to use HTTP Tunneling.

Please answer my both queries.

Original comment by satya.bh...@gmail.com on 11 Dec 2010 at 8:41

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I don't understand why you do step No.1
"Changed talk_base::SocketAddress stun_addr("stun.l.google.com",19302); to  
talk_base::SocketAddress stun_addr("relay.google.com ", 19302)"

The STUN server allows clients to find out their public address, the type of 
NAT they are behind and the internet side port associated by the NAT with a 
particular local port. This information is used to set up UDP communication 
between clients that are apart by NAT and so establish a call.

Original comment by jun...@google.com on 12 Dec 2010 at 6:02

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does this library support TURN? The discussion here 
http://code.google.com/apis/talk/libjingle/important_concepts.html#portssocketsc
onnections suggests not.
I would say much more than 8% of users are behind STUN unreachable  
(institutional) networks.

Original comment by gabriele%mysimpatico.com@gtempaccount.com on 13 Dec 2010 at 1:28

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By Step1 i mean that i am using STUN and TURN both(in case STUN fails behind 
some NATS) using the API 
port_allocator_ =
      new cricket::BasicPortAllocator(network_manager_, stun_addr,
          talk_base::SocketAddress("relay.google.com", 19295), talk_base::SocketAddress("relay.google.com", 19294),
          talk_base::SocketAddress("relay.google.com", 443));

But i don't think your turn is able to bypass the NAT...maybe its not using 
HTTP tunneling and it is using TCP connection? 

Original comment by satya.bh...@gmail.com on 13 Dec 2010 at 3:54

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Libjingle supports a "TURN-like" protocol which is close to an earlier version 
of the TURN. I need to look into the log for further investigation. Could you 
please reproduce the steps you did using "call" with the "-d" switch? meaning 
"call -d" command? then save the screen log and send to me.

Original comment by jun...@google.com on 7 Jan 2011 at 12:34

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I am also facing the same issue. The video and audio(from the file) is not 
reaching other side.

Original comment by krishna....@gmail.com on 12 Jun 2011 at 3:18

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Hi! I am new to VoIP. I want to use Libjingle for my VoIP application. I am 
configured the call client Example with the Relay server for 3rd , 4th, 5th 
parameter in BasicPortAllocator Constructor, but still the client is not 
sending and using Relay candidates.

Could you please guide me how to use the client with the relay server, I 
followed the procedure given in the discussions, but It didn't worked. 

Original comment by niranjan...@gmail.com on 6 Sep 2012 at 5:08