Hi,
Does it also depends on the server that I am using? I am using Asterix.
Is it possible to create a direct RTP session with that or do I have to send
the media only through the server everytime? I guess that it might create an
extra overhead if I pass the media through the server for every call.
With
asterisk you can use the parameter “canreinvite=yes” in sip users configuration,
and asterisk will try to send a re-INVITE when the call is established to allow
the direct end-to-end transmission of the RTP packets.
I have read that when a SIP server like SER receives a message, it can decide
that it wants to stay in the loop or not. If not, SER will provide the
user agents with the information they need to contact each other and then SIP
messages will go directly between the two user agents.Is it possible with
Asterix too?
SER
only processes SIP signalling, not RTP packets. It’s a different issue.
BR,