This actually is a very basic and significant terminology based on which some 
RFCs  are defined unfortunately.

Sec 7 of RFC3388 has clearly indicated the identifier of RTP session as IP+ 
(RTP + RTCP port)

<snip>
   This definition assumes that a single audio (or video) stream maps
   into an RTP session.  The RTP RFC [6] defines an RTP session as
   follows: "For each participant, the session is defined by a
   particular pair of destination transport addresses (one network
   address plus a port pair for RTP and RTCP)".
</snip>

I am not sure if this can be changed easily.

-Rockson

-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Johansson Olle E
Sent: Friday, November 28, 2008 5:51 PM
To: Iñaki Baz Castillo
Cc: [email protected]
Subject: Re: [Sip-implementors] how to identify a RTP session


28 nov 2008 kl. 10.37 skrev Iñaki Baz Castillo:

> El Viernes, 28 de Noviembre de 2008, Rockson Li (zhengyli) escribió:
>> why there're only three RTP sessions, I think it should be six?
>> since "each participant receiving from the other two on separate port 
>> pairs"
>> which means each pariticipant should listen on two separate port 
>> paris for incoming RTP packets, so there're two RTP sessions for each 
>> participant, and should be six for three participants.
>
> I agree. It seems a bug.

I think the terminology section of the RFC needs some explanations.
I would suggest something like.

RTP stream
        RTP + RTCP sent from one part to another

RTP session
         A set of RTP streams for an application. In the case of a SIP
         call between two parties, one RTP stream in each direction,
         meaning a total of two RTP and two RTCP connections.

There are identifiers for the stream, but the session is not very clearly 
defined. The stream IDs are not part of the SDP in a SIP call setup... We just 
have to trust the RTP stream we receive. Or?

/O
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