asterisk users - Oct 2008 - RTP sent before the INVITE ACK (for voicemail app)

Hello,

With asterisk 1.4.11, I am calling AGI "exec voicemail" upon a SIP
INVITE

invite -> asterisk
<- 100
<- 200
<- RTP
ACK ->
...

asterisk is sending the RTP for the greeting before the original invite is
ACK-ed (confirmed with a tcpdump) as if playing the prompt as soon as it is
received from the AGI. I don't see any 183 so I don't think early media
should apply.

CLI output does not show any error that I see. Is there any reason other than a
SIP 183 that would trigger this and isn't asterisk supposed to ACK/answer
the channel before playing any prompt?

Thanks,

Sebastien.
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>
> CLI output does not show any error that I see. Is there any reason other
> than a SIP 183 that would trigger this and isn't asterisk supposed to
> ACK/answer the channel before playing any prompt?
>
Asterisk wil start the audio as soon as it sends back the 200 Ok
response it doesn't wait for the ACK. Most SIP servers will work like
that. The matching of ACK requests to a SIP transaction is not a
particulalrly robust mechanism (for instance if a user agent puts its
IP address in the Call-ID and a SIP ALG fiddles with the SIP packet
for INVITEs but ignores ACKs then there will be a mismatch. This
happens more frequently then you would think) so sending RTP after an
OK response is the correct thing to do.

I think Asterisk will actually cut off the call after 32s if it
doesn't get an ACK which is not such a great idea but that may have
been changed in later versions. The arrival of an RTP packet from the
remote end should be used as the definitive indication of an answered
call not the ACK.

Regards,

Greyman.