[asterisk-dev] Asterisk forwards Audio without early session

Kevin P. Fleming kpfleming at digium.com
Tue Jan 22 11:50:44 CST 2008

Sebastian Damm wrote:

> 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?

No, this is incorrect. Asterisk sent an INVITE to the phone that
included SDP, which means that Asterisk is willing to receive media from
the phone and the phone is free to send it.

The phone's response without SDP means Asterisk should not send audio
*to* the phone, but it does not impact Asterisk receiving audio.

Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

More information about the asterisk-dev mailing list