Subject: [SIP] SDP coder negotiation

From: "Snyder, Duke" <[email-masked]>
Date: Fri, 17 Nov 2000 07:59:19 -0500

In rfc 2543 bis-02, section 16.3, there is a example: 
Bell sends INVITE, sdp=coders 0, 3, 4, 5. 
Watson sends Trying..... 
Watson sends OK, sdp=coders 0 3. 
Bell sends ACK, no sdp. 

How do Bell and Watson determine what coder to use (0 or 3 are common - but
which one) since there is no coder negotiation in RTP.  

Thanks,
Duke

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From: Billy Biggs <[email-masked]>
Date: Fri, 17 Nov 2000 09:47:42 -0600

Snyder, Duke ([email-masked]):

> How do Bell and Watson determine what coder to use (0 or 3 are common
> - but which one) since there is no coder negotiation in RTP.  

  They each choose whichever they prefer.  Bell should be prepared to
receive any of 0, 3, 4 or 5 and Watson should be prepared to receive
either of 0 or 3.

  As well, they should note that the incoming codec can change at any
point in time, and that the incoming RTP can come from any IP address,
not just the one you are sending to.

  The media flow is not a two-way connection, it is two one-way
connections.

-- 
Billy Biggs, 3Com               Email: [email-masked]
http://www.div8.net/billy/      Phone: [email-masked]

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From: "Simon Barber" <[email-masked]>
Date: Fri, 17 Nov 2000 15:29:38 -0800

>   The media flow is not a two-way connection, it is two one-way

They should also be able to receive multiple audio streams, with different
SSRC at a single RTP port. Most implementations out there today do not
conform to the spec and will not handle this. Could I do a quick survey of
implementations out there - what RTP functionality do you support (should
this be in the bake-off torture test?):

1) changing codecs on the fly
2) receiving packets that contain different amounts of audio
3) handling RTCP at all
4) receiving RTP from any address
5) multiple audio streams (how many?)
6) SSRC clash resolution
7) multicast

any others?

Our implementation will support all of the above, and 100s of audio streams.

Simon Barber