One way audio still plaguing WebRTC/Chrome

[James Mortensen]

On the latest Canary (and even on Stable), we're getting occasional one way audio when calling PSTN phones and more often than not, we get 1 way audio whenever we reach a voicemail box.  I was able to replicate on my end and spent an hour gathering as much information as I could possibly think of so that you folks would have the information you need to take a look.  Here's what I looked into myself while on the call.

1.  I didn't see any errors in the chrome_debug.log file.  It looked like STUN binding requests were being sent over and over again to Asterisk, and I believe this is normal.

2.  I could hear audio from the PSTN, but the PSTN phone couldn't hear me.  In other words, Chrome received audio.

3. On the Asterisk 11.5 server, I could see RTP flowing both ways, both to our carrier and from the Chrome client and from the carrier and to the Chrome client.  On Asterisk, it looked like everything was good to go.

4.  I tried to use tshark (TShark 1.6.7) to do an audio capture on the Asterisk server to see if I could hear both audio streams, or at least just the good one; however, the pcap's contain RTP, but Wireshark is unable to extract any VOIP calls.  Here is the tshark command I used:   tshark -i eth0 1way.pcap  I'm not sure if this is because I'm doing this wrong, because Wireshark is configured wrong, or if the SRTP security is on the carrier side as well.  I know the carrier is configured for AVP, so I assume I should be able to capture the audio streams.

5.  I looked to see if I could see re-invites or anything that might send conflicting information to Asterisk/Chrome.  I didn't see anything odd.

I have the Asterisk SIP logs, two pcap files where I tried to unsuccessfully extract audio, a sample of the RTP stream from Asterisk, the Chrome SIP logs, a tcpdump on the client, and screenshots of WebRTC-Internals.  As you can imagine, it's a lot of information.  So let me know what parts you want to see first and I'll include them so I don't overwhelm the system with too many attachments.  

This has happened several times today for several people.  Most of our calls, even from the same number, are two-way audio, so I understand this is a tough problem that's difficult to replicate.

James

[Vikas]

Thanks for the feedback James. It would be good to add these details in the WebRTC issue tracker. The first thing to look into would be webrtc-internals & chrome debug logs.

/Vikas

[James Mortensen]

Thanks Vikas,

I'll create a ticket and attach the internals and the chrome debug logs.  Do you prefer that the issue be created in the Chromium issue tracker https://code.google.com/p/chromium/issues/list or the WebRTC issue tracker?  https://code.google.com/p/webrtc/

Thanks again!

James

James

"Every Call, Every Time!"

How did I do?

Click on a face to provide feedback on my performance!

Synclio Extension:  110627

Phone:  866-707-4590

Skype: No! Call me on Synclio! :)

--
 
---
You received this message because you are subscribed to a topic in the Google Groups "discuss-webrtc" group.
To unsubscribe from this topic, visit https://groups.google.com/d/topic/discuss-webrtc/MOZN1wCLb0g/unsubscribe.
To unsubscribe from this group and all its topics, send an email to discuss-webrt...@googlegroups.com.
For more options, visit https://groups.google.com/groups/opt_out.

[Vikas Marwaha]

James,

It is fine to create it in the webrtc issue tracker.

/Vikas

--
 
---
You received this message because you are subscribed to the Google Groups "discuss-webrtc" group.
To unsubscribe from this group and stop receiving emails from it, send an email to discuss-webrt...@googlegroups.com.

[James Mortensen]

The ticket has been created here:  https://code.google.com/p/webrtc/issues/detail?id=2347&thanks=2347&ts=1378252845

Please let me know if there's anything else I can do to help with this issue.

James