Re: [Sip] Keepalive on media protocols while mute / on-hold etc, & SIP INVITE

Alex,

You are raising two separate issues:

1) how the receiver knows the sender is still there when muted
    or on hold

2) how stupid devices like nats and fws know the stream is still
    active (so the path is preserved) in the same situations

For (1), the answer is supposed to be RTCP. Even when a stream is on 
hold (via SENDONLY or INACTIVE) RTCP is supposed to flow.

But (2) is more difficult. Ideally the stupid devices would be smart 
enough to use the same answer as (1). If they don't, then something will 
need to dribble thru the RTP stream.

The issue of what devices actually do this is another matter altogether.

	Paul

Alex Bligh wrote:
> 100% of the 2(!) vendors I have tested do not appear to send any media
> stream packets unidirectionally (when one end is muted) or bidirectionally
> (re-invite when other party put on hold by UA). This appears not to be in
> conflict with any standard. However, I feel (perhaps incorrectly) a media
> keepalive would be useful so intervening NAT/f/w stuff stays open. Clearly
> this is simple in the mute case (as the muted device can just send whatever
> it likes when it likes), but how about in the re-invite case where the UA
> appears to be being told "no I don't want a media stream". Is there some
> way of saying "but I have to send you a very small keepalive media stream
> or the transient entry in my stateful mid-device will disappear, and please
> don't send me port-unreachables either"?
> 
> The other reason this would be useful is it would allow UAs (including
> media gateways) to reliably respond to lack of media stream (RTP etc.) and
> shut down a hanging SIP session - from what I can see with the UAs I'm
> looking at, one side putting the other on hold, or both ends being on mute,
> is going to tear down a SIP session if this sort of switch is enabled
> without keepalives.
> 
> Alex
> 
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_______________________________________________
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This list is for NEW development of the core SIP Protocol
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sipping@ietf.org for new developments on the application of sip