Dear all,
Does reinvite work for a SIP to SIP call if there are more than one
Asterisk between the clients? An example scenario:
A ---> |Asterisk 1| ---> |Asterisk 2| ---> B
client A behind Asterisk 1 calls client B behind Asterisk 2, after the
connections have been established, the Asterisks issue reinvites and
they will step out of the media path so that RTP traffic will stream
directly between the clients.
If it works, does the protocol between Asterisks need to be SIP or can
it also be IAX2, for example?
I have tested reinvite when the clients are registered to the same
Asterisk server but at the moment I can not test if it works in the
scenario described above.
Thanks,
Mikko Suniala
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