Why not, this is already done when rfc2833 is sent interlaced with voice
why can't interlacing to voice codec be treated in the same. But my
point was that it is not done by many devices so we should not assume
that the peer will be ready to receive any of the codecs in the
supported list.
Cheers
Ramesh
somesh s wrote:
Hello Ramesh,
If that is the case, then let us take up the scenario.
UA1 ====================> UA2
Already 0 & 1 codec have been negotiated between the
two UA. But RTP is being sent with codec 0 from UA1 to
UA2. Let UA1 talk to UA2 for long hours.
But when UA1 suddenly changes the codec from 0 to 1,
without re-INVITE, should UA2 determine the RTP
Payload type and play accordingly?
Suppose if both UA1 and UA2 are mobile, then UA1 never
know how far it is reliable to switch between the
codecs because UA2 may not support the switched codec
now which might had been negotiated earlier?
Am I right?
Thanks in Advance
[EMAIL PROTECTED]
[EMAIL PROTECTED]
--- Ramesh Ramamurthy <[EMAIL PROTECTED]> wrote:
Theoritically you should be ready to receive traffic
in any of the
codecs that was sent out in SDP. There need not be a
reinvite for the
change in Codecs. The reason I say thoritically is
because I have seen
quite a few device that don't support this
functionality, especially
when doing interop with other protocols.
Cheers
Ramesh
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Today's Topics:
1. Doubt about SDP Negotiation (somesh s)
2. New codec in Re-Invite. (Ramesh Ramamurthy)
----------------------------------------------------------------------
Message: 1
Date: Tue, 7 Dec 2004 02:51:30 -0800 (PST)
From: somesh s <[EMAIL PROTECTED]>
Subject: [Sip-implementors] Doubt about SDP
Negotiation
To: SIPimplementors SIP
<[EMAIL PROTECTED]>
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset=us-ascii
Hello All,
I have a question about SDP Negotiation.
Caller A ======== INVITE ===========> Callee B
Caller A sends codecs for example 0 and 1 supported
by
A to B. B sends its codec in 200 OK for example
both 0
and 1.
Now A sends media stream with codec 0 to B.
After some time, if A wants to change the codec to
1
can A do so without RE-INVITE? or if B wants to
change
the codec to 1 can B do so without RE-INVITE?
In short both user agent can switch to any codec
in
codec list got through sdp negotiation without
sending
a new invitation - Is this correct ?,if yes,then
how
can a UA inform that it will not be able to support
a
codec in a codec list got negotiated , in middle of
session )
Thanks in Advance
[EMAIL PROTECTED]
[EMAIL PROTECTED]
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------------------------------
Message: 2
Date: Tue, 07 Dec 2004 09:33:30 -0700
From: Ramesh Ramamurthy <[EMAIL PROTECTED]>
Subject: [Sip-implementors] New codec in Re-Invite.
To: [EMAIL PROTECTED]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1;
format=flowed
Hi,
Can there be an entirely new list of codecs in
reinvite, that is
different from the original capability set sent in
the INVITE that
started the dialog?
Cheers
Ramesh
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