[asterisk-dev] Asterisk forwards Audio without early session

Sebastian Damm damm at sipgate.de
Tue Jan 22 11:05:22 CST 2008


I have seen the following scenario:

Asterisk gets a call from PSTN (via Sangoma PRI card). It sends out an
INVITE to the appropriate SIP peer. The SIP peer answers with 180 without
SDP, but still sends some RTP packets (silent). I don't know why it does so,
but it does. Asterisk now forwards the RTP packets to PSTN, thus does not
generate a Ringing itself. So for the caller the line is silent until the
callee picks up.

According to the developers of the SIP PBX, Asterisk should ignore the RTP
packets, because there is no early media session established. To me, this
sounds correct.

Is this a bug in Asterisk? Or is the behavior desired as it is? If not,
should I raise a bug in bugtracker?

We are using Asterisk 1.4.13 (because there was an issue with querying the
channels via manager on Asterisks with PRI card under high load with 1.4.15),
haven't tested any newer version.

Best regards,
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