[asterisk-dev] Asterisk forwards Audio without early session

[Maksym Sobolyev]

Kevin P. Fleming wrote:
> 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.

I don't think you are correct. If the phone wants to send only it should 
send 183 and use appropriate a=sendonly tag or set media port number to 
0. Otherwise it would be a potential and very big security issue in the 
protocol, when anybody who can intercept the INVITE or detect open UDP 
ports on the UAC could inject his own media stream into the session. Of 
course with NAT detection and so on it's still possible to have this 
issue, but the risk is much lower.

Unfortunately this area is not very well defined in the RFC, so that 
it's question of which of two behaviors makes more sense. But I am 
pretty sure most of the devices on the market won't accept RTP until 
provisional response with SDP arrives.

Regards,
-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
T/F: +1-646-651-1110
Web: http://www.sippysoft.com