Asterisk sip.conf, peer definition: canreinvite option

Versions

Migration from Asterisk 1.2 to 1.4: The "canreinvite" option has changed. canreinvite=yes used to disable re-invites if you had NAT=yes. In 1.4, you need to set canreinvite=nonat to disable re-invites when NAT=yes. This is propably what you want. The settings are now: "yes", "no", "nonat", "update". Please consult sip.conf.sample for detailed information.

canreinvite= was renamed to directmedia= in Asterisk 1.6.2 to more accurately describe what this setting does. See also the closely related setting directrtpsetup.

Asterisk 1.8 added directmediapermit and directmediadeny to limit which peers can send direct media to each other.

Description

This peer option in sip.conf is used to tell the Asterisk server to not issue a reinvite to the client unless really necessary. This is used to interoperate with some (buggy) hardware that crashes if we reinvite, such as the common Cisco ATA 186.

When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other.

• If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.
• If the clients use different codecs, Asterisk will not issue a re-invite.
• If the Dial() command contains ''t'', ''T", "h", "H", "w", "W" or "L" (with multiple arguments) Asterisk will not issue a re-invite.

'canreinvite=no' stops the sending of the (re)INVITEs once the call is established. From messages in the archives and the Asterisk handbook one finds out that the Cisco ATA-186 does not handle the (re)INVITE well. This is necessary if the client and the Asterisk server is on opposite sides of a NAT gateway or firewall.

• canreinvite = yes "allow RTP media direct"
• canreinvite = no "deny re-invites"
• canreinvite = nonat "allow reinvite when local, deny reinvite when NAT"
• canreinvite = update "use UPDATE instead of INVITE"
• canreinvite = update,nonat "use UPDATE when local, deny when NAT"

Note: In spite of its name 'canreinvite' being set to 'no' does *NOT* disable all reINVITE operations. It *only* controls Asterisk generating reINVITEs for the specific purpose of setting up a direct media path. If a reINVITE is needed to switch a media stream to inactive (when placed on hold) or to T.38, it will still be done, regardless of this setting!

Notes

• reinvite=yes/no is plain wrong, even if you see it mentioned in example .conf files. The correct syntax is canreinvite=yes/no
• Connecting media paths direct to an endpoint behind NAT won't be pretty. Especially if both devices are behind NAT. You might want to try using SER's nathelper in conjunction since nathelper.so can rewrite the SDP so that the private IP addresses are not included in the re-invite.
• When dtmfmode=rfc2833, asterisk will send the RTP stream through asterisk. With dtmfmode=info canreinvite works properly.
• Asterisk 1.8 added the media_address= configuration option which can be used to explicitly specify the IP address to use in the SDP for media (audio, video, and text) streams.

Background info

In general Asterisk supports 3 methods of media handling:

1. ...