Subject: [SIP] audio frame length

From: Chris Durand <[email-masked]>
Date: Wed, 30 Aug 2000 15:57:30 -0600

Does SIP have a mean of agreeing on an audio frame length betweem two
devices (e.g .30ms of audio exchanged in each RTP packet).
Thanks

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From: "Umesh Sirsiwal" <[email-masked]>
Date: Wed, 30 Aug 2000 18:49:34 -0400

Hello,
I am interested in knowing if there is anyone who is working 
on a SIP implementation for Mobile/Wireless applications.

-Umesh

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From: Christer Holmberg <[email-masked]>
Date: Thu, 31 Aug 2000 11:09:23 +0300

Hi,

What you have is the a=ptime SDP parameter. However, as the draft says,
it is only used as a "recommendation", and that it should not be
necessary to know ptime to handle the media stream.

Regards,

Christer Holmberg
Ericsson Finland

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From: Colin Perkins <[email-masked]>
Date: Thu, 31 Aug 2000 08:42:44 +0100

>devices (e.g .30ms of audio exchanged in each RTP packet).

a=ptime: in SDP.

Colin

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From: Jonathan Rosenberg <[email-masked]>
Date: Thu, 31 Aug 2000 11:19:24 -0400

Actually, the RTP spec says you do not need to know ptime. From rfc1890:

>  For packetized audio, the default packetization interval should have
>    a duration of 20 ms, unless otherwise noted when describing the
>    encoding. The packetization interval determines the minimum end-to-
>    end delay; longer packets introduce less header overhead but higher
>    delay and make packet loss more noticeable. For non-interactive
>    applications such as lectures or links with severe bandwidth
>    constraints, a higher packetization delay may be appropriate. A
>    receiver should accept packets representing between 0 and 200 ms of
>    audio data. This restriction allows reasonable buffer sizing for the
>    receiver.
> 

Notice the next to last sentence.

This has been expanded upon in the revised spec:

> For packetized audio, the default packetization interval SHOULD have
>    a duration of 20 ms or one frame, whichever is longer, unless
>    otherwise noted in Table 1 (column "ms/packet").  The packetization
>    interval determines the minimum end-to-end delay; longer packets
>    introduce less header overhead but higher delay and make packet loss
>    more noticeable. For non-interactive applications such as lectures or
>    for links with severe bandwidth constraints, a higher packetization
>    delay MAY be used.  A receiver SHOULD accept packets representing
>    between 0 and 200 ms of audio data. (For framed audio encodings, a
>    receiver SHOULD accept packets with a number of frames equal to 200
>    ms divided by the frame duration, rounded up.) This restriction
>    allows reasonable buffer sizing for the receiver.

Same basic idea.

It is for this reason that ptime is not the subject of "negotiation";
its an fyi parameterm.

-Jonathan R.

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