Subject: [AVT] Multicast RTP session.

From: B Prashanth <[email-masked]>
Date: Fri, 28 Feb 2003 15:03:29 +0530

Hi
I am new in this list and am working on developing application layer 
stacks for streaming and telephony application. I had a very fundamental 
question in this regard. How does RTP handle incoming packets coming 
from number of sources on a multicast port? Is the
1. Are the packets (coming from different SSRCs) given to the 
application using the stack in the order of their arrival.
OR,
2. Seperate jitter buffers would be needed to be maintained in the RTP 
stack for each of the SSRCs (senders of RTP data) so that the jitter 
from each of the senders is removed individually and then the payload be 
given to the application that is using this stack.
Thanks in advance,

-- 
Prashanth 

_______________________________________________
Audio/Video Transport Working Group
[email-masked]
https://www1.ietf.org/mailman/listinfo/avt

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From: Colin Perkins <[email-masked]>
Date: Fri, 28 Feb 2003 10:01:41 -0500

Hi,

>application using the stack in the order of their arrival.

This is a valid approach, if the jitter buffer and playout code is part of
the application. The RTP library I wrote as part of the Robust-Audio Tool 
(http://www-mice.cs.ucl.ac.uk/multimedia/software/rat/) works like this. 
It gives you a very simple RTP library, leaving a lot to the application.

>given to the application that is using this stack.

This is a valid approach, if the playout buffer is managed by the RTP
library. The resulting application will be simple, wrapped around a
more complex playout framework provided by the RTP library.

Colin