[asterisk-users] SIP Question - Early audio one-way or 2-way?

[Faisal Hanif]

You can create trunk/route specific dial command parameters.

Regards,

Faisal Hanif

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steve Davies
Sent: Friday, August 24, 2012 8:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

On 24 August 2012 15:34, Faisal Hanif <faisal at vopium.com> wrote:
> Steve Davies <davies147 at gmail.com> wrote:
>>Hi SIP Gurus,
>>
>>I've tried to find the relevant RFCs, but am struggling. I can find 
>>the odd opinion online, but was wondering if anyone could give a 
>>definitive answer.
>>
>>If a SIP call is initiated (INVITE) and receives either a "180 with 
>>SDP", or a "183 with SDP", then the remote party will start to send 
>>audio for the inband-ringing. Asterisk then passes this audio, and it 
>>is correctly heard by the caller.
>>
>>At present, Asterisk will also start to pass back any handset audio in 
>>return, in theory allowing a conversation to occur on an unanswered 
>>channel if an endpoint were designed to allow this (free phonecalls 
>>here we come!).
>>
>>My question:
>>
>>Should:
>>1) Asterisk block outbound audio between the 183 Progress and the 200 
>>OK packets?
>>2) Replace any outbound audio with silence?
>>3) Replace outbound audio with a special NULL RTP of some sort? Does that
exist?
>>4) Allow any audio to be sent regardless?
>>
>>I have implemented 1) at present on our test rig, but the lack of 
>>outbound RTP causes issues with firewall state not being set-up to 
>>allow the inbound audio. I am not sure how hard/easy it would be to do
>>2) as you'd need to create silence of the correct duration to replace 
>>each audio frame.
>>
>>Thoughts please?
>>
>>Many thanks,
>>Steve
>>
> hi,
>
> you can simply avoid this by using local ring r option in dial 
> command. azterisk pass local ring voice to caller and will not bridge 
> b leg audio until b leg is answered.iin Regards,
>
> Faisal Hanif
> (sent from phone)

Nice thought, but what if there is a useful reason for the progress audio?
If it is sent we want to hono[u]r it, and if it is not, we expect a "180
ringing", and let the SIP device generate the tone, rather than send an
unwanted audio stream to use up bandwidth :)

For example, some UK ISDN services will give a useful call failure message
by sending a progress-frame, followed by some audio. DAHDI and SIP handle
this nicely  with a 183 progress message, and pass on the message on the
un-answered SIP channel.

Regards,
Steve

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