[Live-devel] liveMedia appropriate?

[Dan Stanfill]

>> Basically, I am adding a server to the voice chat used in games. Since  
>> the
>> server will be remote, I am assuming that it is going to want to both  
>> send
>> and receive unicast packets, not multicast.
>
> What do your clients (i.e., the games) currently do?  Do they use TCP or  
> UDP, and if they use UDP, do they use RTP-over-UDP?
>
> Or have your clients (the games) not been designed/written yet?

Existing games are using a proprietary 3rd party library which is strictly  
peer-to-peer and tied in closely with the rest of the game traffic. A  
quick implementation using this library demonstrated we would be better  
doing our own transport for a variety of reasons.

Underneath, this library uses UDP without RTP.

> Assuming that your clients are using RTP (see above), then the "LIVE.COM  
> Streaming Media" code would seem to be a good match, although there  
> would be quite a bit of programming that you would still need to do.   
> Note, in particular, that the LIVE.COM libraries don't include any codec  
> (i.e., audio/video encoding/decoding/transcoding functionality), so that  
> would need to be added.

We are not using RTP, and initially I was thinking just raw UDP would be  
fine, but it seems that an RTP implementation might speed us up by  
handling a lot of the low-level bookkeeping, such as packet  
reordering/discarding, etc. Also, RTCP looks like it might provide useful  
info back for adjusting network usage dynamically.

> The "testRelay" demo application (see  
> <http://www.live.com/liveMedia/#testProgs>) is an example of a simple  
> relaying application.

One thing that wasn't clear to me in testRelay was the best way to  
elegantly handle the multiple clients and one-to-many mapping for each  
client.

Thanks for the feedback,

Dan


-- 
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Dan Stanfill
Pinniped Software
http://www.pinniped.com
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