[asterisk-users] RTP sent before the INVITE ACK (for voicemail app)

[tic tac]

Thanks, in my case though it looks like the originating party (polycom softphone) is hearing a clipped voicemail prompt because of that; should I look into having that fixed into their firmware? As a workaround, I was thinking to just add a few seconds delay in app_voicemail, or wait through AGI before calling voicemail, makes sense?



> Date: Wed, 1 Oct 2008 15:43:37 +0100
> From: greymanvoip at gmail.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] RTP sent before the INVITE ACK (for voicemail	app)
> 
> >
> > 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.
> 
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