Hi all!
I have been looking around for hardware to place in a PSTN/SIP gateway. For
RTP, it looks like for all hw, you have to specify which codec should be
used for a specific "channel" you setup (for the RTP to be "translated" to
PSTN and the other way around). Now I wonder what such a GW should offer in
SDP.
Imagine this case:
PSTN subscriber calls SIP subscriber through a PSTN GW. When the gw receives
the call on pstn it has to send a INVITE to the SIP network and include sdp.
What is most correct of the two following cases:
1) The gw send a the complete list of codecs it supports, say 0,8,18. The UA
called can handle the same codecs and therefore answers with 0,8,18. The GW
must setup the connection between ip and pstn and, as stated above, must
start a "channel" and set the codec to use. It choses the first one (0).
According to this setup, should the gw be able to receive ANY of the codecs
in the response? Say UA starts sending using 0, everything is ok and then UA
decides to send using 8 instead. Is it allowed to do this without sending
reInvite/Update? If it does, the gw will not handle voice good at all
2) gw has a priorization of codecs, This means that it will only offer the
codec with the highest prio in the sdp, say 8, so it only includes codec 8
in the sdp offer. This ensures that in the response back there will only be
one codec and the gw can start the "channel" without having to worry about
UA will start sending other formats without sending reInvite/Update. The
risk here is that the UA wotn support the codec that GW has as higest prio
and therefore no call will be set up.
Is my assumptinos above correct?
If so, which of the two alternatives above do you recommend?
Regards,
// Andreas
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