G711u Audio issue at the Chrome browser

[Sandeep K S]

Hello Experts,

 I am writing a small application to do SRTP-RTP interworking between a webrtc Client on Chrome, and a softclient running on some OS like Mac, or windows.

I am facing a very strange problem, and not able to understand the reason for the problem. So need your expert opinion on this.

I am initiating a call from the webrtc app on chrome to a softclient via my application. 

There is clear audio from browser to my softclient. However, there is no audio from the soft client towards the browser.

Chrome version: 28 (tried with 31 as well, and same issue was seen)

Troubleshooting tried:

1. We captured packets using wireshark on the softclient side and the browser side. And we can clearly see that the SRTP packets are flowing towards the browser, and RTP packets are flowing towards the soft client.

2. We checked the ICE connectivity checks also. That is happening for all ports.

3. We looked at the webrtc-internals, and it was clearly showing the packets recieved incrementing. Also, the statistics graph for packets received was showing around 70-80Kbps.

we then tried a different soft client, and had 2 way audio with this different client. No changes made to my application. But it worked for this client, not for the other client i tried earlier.

So want to know what could be the reason why the browser was not able to play out the SRTP PCMU stream out. I looked at the chrome_debug.log file and couldn't see any difference. No error logs seen related to the SRTP PCMU stream also.

So want your help in debugging this issue further. I specifically need some help on how i can debug this on the browser. I need to know what is going on wrong which probably needs some correction either at the browser end or at the softclient end.

Regards

Sandeep K S

[Kaiduan Xie]

You need to turn on webrtc media related log to see what is going on with the srtp packets.

/Kaiduan

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[Sandeep K S]

Can you please let me know how to turn on the webrtc media related logs?

Currently, i am running google-chrome with --enable-logging -v=11 parameters

-Sandeep K S

[Kaiduan Xie]

Please do a test with libjingle log on first,

--enable-logging --vmodule=*third_party/libjingle/*=3

If you still can not find out the root cause, and you need webrtc media related log, you have to build chrome source code by following the link below,

https://groups.google.com/forum/#!topic/discuss-webrtc/4YcHasYC3xg

/Kaiduan

[Sandeep K S]

Hello Kaiduan,

Thanks for the help. 

From the logs, now I see the below difference. 

For the non-working case, there is a failure message seen as below:

"[1119/113235:VERBOSE2:webrtcvoiceengine.cc(2261)] SetOutputVolumePan(0, 1, 1) failed, err=8040"

I do not see this message for the working case. So could this be the cause for the audio issue? Also, when i looked into the code for the SetOutputVolumePan, I see that this is checking if the audio device supports Stereo or not. What I do not understand is that I have the same device on which I run the broswer for the webrtc call. However, just changing the softclient on the other end either fails in this OutputVolumePan routine or does not. So can you shed some more light on this please?

-Sandeep

[Justin Uberti]

This sounds like something we should check out. Can you file a bug on this error?