June 2005
Intermediate to advanced
502 pages
21h 48m
English
Content preview from Switching to VoIP
Become an O’Reilly member and get unlimited access to this title plus top books and audiobooks from O’Reilly and nearly 200 top publishers, thousands of courses curated by job role, 150+ live events each month,
O’Reilly covers everything we've got, with content to help us build a world-class technology community, upgrade the capabilities and competencies of our teams, and improve overall team performance as well as their engagement.I wanted to learn C and C++, but it didn't click for me until I picked up an O'Reilly book. When I went on the O’Reilly platform, I was astonished to find all the books there, plus live events and sandboxes so you could play around with the technology.I’ve been on the O’Reilly platform for more than eight years. I use a couple of learning platforms, but I'm on O'Reilly more than anybody else. When you're there, you start learning. I'm never disappointed.I'm always learning. So when I got on to O'Reilly, I was like a kid in a candy store. There are playlists. There are answers. There's on-demand training. It's worth its weight in gold, in terms of what it allows me to do.
This is the Title of the Book, eMatter Edition
Copyright © 2007 O’Reilly & Associates, Inc. All rights reserved.
128
|
Chapter 6: Replacing the Voice Circuit with VoIP
Per-peer codec selection on Asterisk
In Asterisk’s sip.conf configuration, it’s possible to set an order for codec preferences
on a per-peer basis. Consider the following snippet of a sip.conf file:
[103]
type=friend
context=default
username=103
callerid=103 Budgetone Phone
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
This definition, for a SIP peer called 103, allows only G.711 µlaw and GSM codecs to
be employed. Asterisk prefers the codecs in the same order specified in each SIP peer
definition.
What this sample SIP peer cannot do is have an independent call path. In order to
allow SIP peers to operate with independent call paths, the
canreinvite setting must
be
yes. This enables SIP phones to establish a direct media channel with the phone
on the other end of the call, cutting the softPBX out of the proverbial loop. Enabling
this option doesn’t force the SIP phones to use an independent call path, though: in
situations in which transcoding is needed or the softPBX is required in the call path
because of the design of the network, Reinvite won’t get used.
Key Issues: Replacing the Voice Circuit
with VoIP
• The two fundamental duties of a softPBX are call signaling and voice transmission.
• Voice channels in a VoIP network are analogous
Become an O’Reilly member and get unlimited access to this title plus top books and audiobooks from O’Reilly and nearly 200 top publishers, thousands of courses curated by job role, 150+ live events each month,
and much more.
Read now
Unlock full access
More than 5,000 organizations count on O’Reilly
O’Reilly covers everything we've got, with content to help us build a world-class technology community, upgrade the capabilities and competencies of our teams, and improve overall team performance as well as their engagement.Julian F.
I wanted to learn C and C++, but it didn't click for me until I picked up an O'Reilly book. When I went on the O’Reilly platform, I was astonished to find all the books there, plus live events and sandboxes so you could play around with the technology.Addison B.
I’ve been on the O’Reilly platform for more than eight years. I use a couple of learning platforms, but I'm on O'Reilly more than anybody else. When you're there, you start learning. I'm never disappointed.Amir M.
I'm always learning. So when I got on to O'Reilly, I was like a kid in a candy store. There are playlists. There are answers. There's on-demand training. It's worth its weight in gold, in terms of what it allows me to do.