[Asterisk-Dev] Hello Everybody

[Jerris, Michael MI]

Sorry about the previous message.  All changes to asterisk go through
the bugtracker at bugs.digium.com.  Please read the bug guidelines on
that site for more details.   


	                         I just wished to know that in VoIP,
whether the RTP and SIP protocols are modified according to the needs.
If yes, for what reasons are the changes are made and where are the
changes done. And it would be also very kind of you if you could let me
know how do I go about doing the same. Kindly reply.

		Regards,

		Bharat M. Sarvan

		 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-dev/attachments/20050519/3d006ffd/attachment.htm