[gst-devel] GStreamer for use in telephony

[Brian Fahrlander]

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!

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

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Brian Fahrlander                             Linux Zealot, Conservative,
Chicago, IL                                       and Stranded Technomad
ICQ 5119262                       http://www.kamakiriad.com/aboutme.html
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