[vorbis] Vorbis over RTP

[Roland Parviainen]

Marshall Eubanks wrote:
> 
> Dear Roland;
> 
> I am working on this off and on in the context of the free amp player.
> We hope to begin multicast streaming of ogg/vorbis early in the new year.
> There needs to be a RFC written for the ogg/vorbis
> data payload format so it can become a RTP type. I was hoping to
> get this done in time of the next IETF in San Diego, but it is too late for
> that. I definitely want to do this in time for
> the Spring 2001 IETF in Minneapolis in March.
> 
> With multicast streaming, packets can be dropped at any time, so
> it is useful to keep each packet as autonomous as possible (each
> packet should ideally be an application data unit, or ADU, as
> Ross Finlayson did for the MP3 format - see
> http://www.ietf.org/internet-drafts/draft-ietf-avt-rtp-mp3-04.txt ).

I know, I've implemented version 2 (i think) of that draft, but 
haven't had time to update my implementation to the latest one
yet. 

> In your RTP, do you send out the comments and codebook as a separate
> packet, or as part of a data packet ? Can these change in mid-stream ?
> It would also be nice if each packet was kept under 576 bytes (or
> at least 1500 bytes) in size, including the IP/UDP/RTP headers,
> to avoid fragmentation. Do you worry about packet sizes in your
> implementation ?

I send four different types of packets: data, info, comment and 
codebook. I dont' care about the size of data packets since they are 
usually less than 500 bytes, but that's just because I'm lazy :). 
The other packets are fragmented to about 1400 bytes (1400 + headers) 
if they are larger than that. The codebook is usually 3 or 4 fragments.

The payload header also contains a 8 bit id that changes when client 
have to reset the decoder with a new codebook, so there is no problem
with sending e.g. a 128 kbit/s file followed by a 160 kbit/s file,
except 
for a small delay.

//Roland Parviainen

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