[gst-devel] GStreamer for use in telephony

zaheer at grid9.net zaheer at grid9.net
Tue Mar 6 02:42:10 CET 2001

On Mon, 5 Mar 2001, Brian Fahrlander wrote:

> zaheer at grid9.net wrote:
> > 
> > Hi
> > 
> > I'd like to introduce myself.  My name is Zaheer Merali, and I am a
> > developer on the PreViking project.  PreViking is open source
> > telephony middleware that allows telephony applications to be 
> > written easily and providing facilities to these applications such as
> > intelligent call routing, audio prompt management, audio switching all
> > on top of multiple telephony hardware (including lets say VOIP stacks
> > such as H.323 and SIP).
> > 
> > Having read the interview on linux.com, I have become interested in
> > the possible use of GStreamer in our middleware.  What we have been
> > looking at was to stream audio from a URL (local file or remote) to a
> > driver process using RTP.  The drivers support limited codecs, mainly
> > just G.711 mu law.  We already have some code to do the streaming of
> > just mu law G.711 audio now, but by using GStreamer it seems that we
> > can add many more formats in a more elegant manner.
>     Oh, wow...first the great, lucid, technical discussion of this beast
> and where it's headed, and this this message that tells me I'm not the
> only one interested in using it for telephony...the little hairs on the
> back of my neck are standing up!

That's good so we aren't alone :)

>     When you guys get some code, let me know- I'll beta test it for ya.

We have a stable release of PreViking, without any of the media conversion
or RTP streaming but is pretty powerful from an IVR and switching
department.  It can be downloaded from


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