[gst-devel] GStreamer for use in telephony

zaheer at grid9.net zaheer at grid9.net
Tue Mar 6 02:18:39 CET 2001


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.

Would GStreamer be the right kind of library to use in this scenario.  The
difference that I can see is that the audio is sent to a telephone device
rather than to a soundcard....


Zaheer Merali
PreViking - open source telephony middleware

More information about the gstreamer-devel mailing list