[asterisk-users] ignore 183 session progress in parallel call scenarios

[Hristo Trendev]

Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are not passed back to the caller.

This behavior is completely correct, because there is no way to know which
early media audio stream to pass back to the caller in a parallel call
scenario (as in this case several endpoint may indicate session progress
all at the same time).

The question is why is asterisk still sending 183 session progress back to
the caller if no audio is to be bridged before the 200 OK anyway? If 183
are not passed back to the caller, then at least a 180 Ringing that may
come from another endpoint will cause the calling endpoint to generate
local ringback. This won't happen if the caller has received a 183 already.

So it's a bit of a race condition as well - if the first endpoint to reply
sends a "183 session progress" this means the caller will not hear any
ringback even if some of the other endpoints are sending back 180 Ringing.

The question is can I somehow block 183 messages from being passed back to
the calling endpoint when dialing several destinations in parallel? I don't
see a point (please correct me if I'm wrong) to pass only the 183 SIP
message back to the caller without the corresponding RTP stream, so it may
be much better to actually ignore it when dealing with parallel call
scenarios (bug?).

BR,
Hristo
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