Subject: RE: [rohc] RE: [Sip] New I-D on compression of SIP

From: "Christian Huitema" <[email-masked]>
Date: Wed, 25 Jul 2001 09:00:10 -0700

> Assume a 9.6kbps single-channel GPRS link.
> 
> Assume that approximately 5kbps of that is used by a 
> half-rate AMR (Alternative Minimum Rate) codec. That gives us 
> roughly 4.6kbps available for processing signaling, such as a 
> re-INVITE, BYE, or new INVITE arriving.

Dean, let's do a reality check. We are constantly receiving requests to
tweak our TCP stack in order to match the long delays of GPRS channels
-- more than 600 ms is not uncommon. These delays seem to be intrinsic
to the technology, probably due to a mix of contention on access and low
level retransmission. Do you really believe that you are going to
conduct a meaningful VoIP conversation over this kind of channel?

-- Christian Huitema

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From: "Dean Willis" <[email-masked]>
Date: Thu, 26 Jul 2001 16:05:00 -0500

Store and forward speech messaging delivered on a one-to-many basis, aka
"group voice chat" is a potential application over high-latency networks
like GPRS/GERAN. Real-time two-way may be problematic for the first
generation of systems, but GPRS/UTRAN should be much better. Maybe the real
wireless wonks (I'm not one!) can shed some light . . .

And I think the latency problem is derived from the way that TDM applies,
wherein the mobile waits for a request slot, requests a slot to talk in,
waits for a response telling it which slot to use, and then waits for the
slot to occur. I wouldn't be surprised if somebody comes up with an
isochronous model akin to the DOCSYS "unsolicited grant" that will work
nicely for VoIP.

--
Dean