Switching to VoIP

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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