asterisk users - Aug 2012 - SIP Question - Early audio one-way or 2-way?

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)

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
>
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24 aug 2012 kl. 16:18 skrev Steve Davies <davies147 at gmail.com>:
> 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?
First, because of Asterisk's RTP implementation we have to send some RTP
packets at this point. You could mute the calling channel before calling and
unmute the channel at answer if needed, but normally sending audio won't
hurt. A normal user should not be able to send early media on a pstn-like
installation where you bill the calls, so there should be little risc of two-way
conversations before an answer.

In some cases you have to let the caller send DTMF (the famous fed ex example)
in
early media, so we can't block any media by default in Asterisk.

Using the "r" option in dial causes a lot of issues, since you can
still get busy or congestion when you have early media, so that is not a good
solution.

/Olle

* The New Edvina SIP Masterclass - Stockholm and Miami this fall!
http://edvina.net/

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