Re: [asterisk-users] Ringing (180) no SDP to progress(183) with SDP transition => no audio.

2017-11-20 Thread Richard Mudgett
On Mon, Nov 20, 2017 at 7:31 AM, Benoit Panizzon 
wrote:

> Dear List
>
> I am testing various early audio scenarios with different voice IC's,
> phones and pbxes.
>
> In Switzerland, when you operate a value added number, you have to
> announce the price of the call, usually in early audio, before the call
> is established.
>
> In 'dialplan' terms this would be:
>
> exten => XX,1,Ringing
> exten => XX,n,Wait(15)
> exten => XX,n,Progress
> exten => XX,n,Playback(price-announce,noanswer)
> exten => XX,n,Wait(5)
> exten => XX,n,Answer
>
> I see the asterisk playing the early announcement audio in the rtp
> stream. Some devices (arris EMTA) calling the asterisk also do play it
> to the caller.
>
> But!
>
> Most other devices I have tested just keep playing the locally generated
> ringtone despite getting an 183 with SDP and the announcement is never
> to be heard by the caller.
>
> If I do to force inband ringback tone, this works with all devices I
> have tested so far.
>
> exten => XX,1,Progress
> exten => XX,n,Ringing
> exten => XX,n,Wait(15)
> exten => XX,n,Playback(price-announce,noanswer)
> exten => XX,n,Wait(5)
> exten => XX,n,Answer
>
> Is anything wrong with the transition of ringing without SDP (to have
> the local device generating ringback tone) and then start sending early
> audio with 183?
>

Both orderings of Ringing and Progress are valid.  It is up to the calling
device to handle it.  As you have seen, there is quite a difference in
how devices handle it.  I have even seen where the calling device needs
Ringing before Progress to handle the call correctly.  I think that case was
because the device was converting ISDN to SIP.  I do think that the devices
that don't stop local ringback in favor of the incoming RTP stream following
the 183 are broken.  Unfortunately it is something that is out of your
control.

Richard
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Ringing (180) no SDP to progress(183) with SDP transition => no audio.

2017-11-20 Thread Benoit Panizzon
Dear List

I am testing various early audio scenarios with different voice IC's,
phones and pbxes.

In Switzerland, when you operate a value added number, you have to
announce the price of the call, usually in early audio, before the call
is established.

In 'dialplan' terms this would be:

exten => XX,1,Ringing
exten => XX,n,Wait(15)   
exten => XX,n,Progress
exten => XX,n,Playback(price-announce,noanswer)
exten => XX,n,Wait(5)
exten => XX,n,Answer 

I see the asterisk playing the early announcement audio in the rtp
stream. Some devices (arris EMTA) calling the asterisk also do play it
to the caller.

But!

Most other devices I have tested just keep playing the locally generated
ringtone despite getting an 183 with SDP and the announcement is never
to be heard by the caller.

If I do to force inband ringback tone, this works with all devices I
have tested so far.

exten => XX,1,Progress
exten => XX,n,Ringing
exten => XX,n,Wait(15)   
exten => XX,n,Playback(price-announce,noanswer)
exten => XX,n,Wait(5)
exten => XX,n,Answer

Is anything wrong with the transition of ringing without SDP (to have
the local device generating ringback tone) and then start sending early
audio with 183?

-BenoƮt Panizzon-
-- 
I m p r o W a r e   A G-Leiter Commerce Kunden
__

Zurlindenstrasse 29 Tel  +41 61 826 93 00
CH-4133 PrattelnFax  +41 61 826 93 01
Schweiz Web  http://www.imp.ch
__

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users