[Asterisk-Dev] FW: [Asterisk-Users] RTP timing issues

[Bart Coppens]

Dear Sirs,


The Asterisk bounty has been updated accordingly.

Some info about our environment:

Our Asterisk server is logically connected to a Veraz NGN platform
through SIP and we are facing two major problems for calls from/to
Veraz;

When calling from Veraz to any SIP extension, no ringback is generated
as Veraz does not generate any RTP packets until Answer supervision.
Asterisk can not deliver ringback.

Calling to Veraz is problematic as all our interfaces are using Silence
compression.


>-----Original Message-----
>From: steve at daviesfam.org [mailto:steve at daviesfam.org]
>Sent: Thursday, October 07, 2004 11:08 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Cc: asterisk-dev at lists.digium.com; Bart Coppens
>Subject: Re: [Asterisk-Users] RTP timing issues
>
>On Thu, 7 Oct 2004, Bart Coppens wrote:
>
> > Some time ago, I announced a bounty to solve the issues with regards
>to
> > silence compression (chopped voice) and one way voice. To get this
>solved,
> > Asterisk should get the clocking from an internal source in a way that
>an
> > ouput stream can be generated without getting any RTP input.
> >
> > Now my company is exposing a payment of 1000USD for this bounty. This
> > payment have to justified through an official invoice.
> >
> > Can someone give me an indication if this can be achieved?
>
>
>It can be achieved.
>
>Steve
>

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