problem with synchronization between multiple audio clients

[Mario Montagud Climent]

Dear Javier,

I am also interested in the development of a system as the one you  
described in your mail, using RTP/RTCP and GStreamer.

Answering to your question, the specification of the RTP/RTCP (RFC  
3550) does not support the functionality you are interested in.  
However, our research group has designed an Inter-Destination Media  
Synchronization (IDMS) solution, which is based on the use and  
extension of RTP/RTCP protocols, and is being standardized within the  
IETF (see https://datatracker.ietf.org/doc/draft-ietf-avtcore-idms/).  
This IDMS solution is targeted to enable synchronized playout across  
geographically distributed rendering platforms.

I implemented this IDMS solution in Network Simulator 2 (NS-2). Now, I  
also want to do so in Gstreamer to be able to conduct user perception  
(QoE) tests in real scenarios.

If you want to know more details about our IDMS solution, you can  
check our upcoming IETF standard and the following recent papers:

[Mon13] M. Montagud, F. Boronat, H. Stokking, Early Event-Driven (EED)  
RTCP Feedback for Rapid IDMS, The 21st ACM International Conference on  
Multimedia (ACM MM 2013), Barcelona (Spain), 2013 (October 21-25).

[Mon12]  M. Montagud, F. Boronat, H. Stokking, R. van Brandenburg,  
Inter-Destination Multimedia Synchronization; Schemes, Use Cases and  
Standardization, Multimedia Systems Journal (Springer), 18(6),  
459-482, November 2012.

I posted a mail to this list a few months ago, but I was waiting for  
the release of GStreamer 1.2.0 to begin with the implementation tasks.  
The main reason was, as mentioned by Tim, the new features that have  
been recently added to gst-rtsp-server: multicast support and  
distribution of clock sync info (both base time and time provider to  
sync to). These two functionalities are very important for enabling  
IDMS.

Also, the videos of the talks by Win Taymans and Edward Hervey will be  
very useful to help understanding these concepts. Btw, will the videos  
of the GStreamer Conference 2013 be posted soon?

Recently, Thomas Roos and Win Taymans added a patch to solve an issue  
regarding the RTCP timing rules and the behavior of clients when  
receiving the first RTCP packet. The implementation of the RFC6051  
(Rapid Synchronisation of RTP Flows) was also planned for this year.  
Both issues are also important for IDMS!

To my understanding, other important aspects are: the jitter buffer  
configuration, the ts-offset attribute, and the sink method.

I plan to begin the implementation tasks next month, according to the  
schedule of my PhD thesis.

Hope this helps!

Cheers,

Mario Montagud

Javier Domingo <javier.domingo at fon.com> escribió:

> Hi Tim!,
>
> Thanks a lot for the info! I was just wondering whether if the RTP[1]
> standard would be capable of that. I had already seen aurena, but I
> prefer the RTP solution. I will be doing experiments with it though,
> and report back if I encounter something interesting,
>
> Cheers,
>
> Javier Domingo Cansino
> Research & Development Junior Engineer
> Fon Labs Workgroup, Getxo - Spain
>
> [1] RTP Standard: http://tools.ietf.org/html/rfc3550
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