Re: [AVT] 2833bis - RTP ping

[Henning Schulzrinne <hgs@cs.columbia.edu>]

Continuing on Flemming's comments:

> Other:
> =====
> RTP continuity:     Continuity tests are being performed in the PSTN 
> to verify
> the operation of the trunk over which your call is setup. It turns 
> out, that
> it's very useful to have a similar capability to verify continuity of 
> an RTP
> session. While RTCP does offer some help in this area, it's not a complete
> solution for a couple of reasons; 1) the RTP and RTCP destination 
> ports differ,
> 2) you can't force the other party to actually send you an RTCP packet 
> when you
> would like to (e.g. immediately to verify RTP continuity on call 
> setup). If we
> had something like a "ping" and a "pong" event, it would be very 
> helpful (if
> you get a "ping", you send a "pong"). Comments ?
Clearly, this would only make work if you knew ahead of time whether the 
other side supports this feature. I can see the usefulness of this 
feature, but there are some details that may make this more than just 
another special event. A few questions:

(1) Who should send this? Clearly, end systems, but what about RTP 
mixers and translators?

(2) If mixers just replicate the packet without knowledge of the 
content, they'll get a response from each participant. Is that ok?

(3) What happens if you send this to a multicast address with 10,000 
participants? The sender and receiver may not actually know that this is 
multicast, since there may be a translator that bridges between unicast 
and multicast. The translator may also be ignorant of the special 2833 
event type (and 2833 in general), but will quickly figure out that it 
did something special, after getting flooded with 10,000 "pong" packets...

With RTCP, we have rate-limiting options in place which deal with these 
issues, particularly (3).


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