SIP trunk between CME and CM

Posted by Andy on May 5, 2009

Best Practices

Follow these guidelines :

•Configure a SIP Trunk Security Profile with Accept Replaces Header selected.

•Configure a SIP trunk on Unified CM using the SIP Trunk Security Profile created, and also specify a ReRouting CSS. The ReRouting CSS is used to determine where a SIP user (transferor) can refer another user (transferee) to a third user (transfer target) and which features a SIP user can invoke using the SIP 302 Redirection Response and INVITE with Replaces.

•For SIP trunks there is no need to enable the use of media termination points (MTPs) when using SCCP endpoints on Unified CME. However, SIP endpoints on Unified CME require the use of media termination points on Unified CM to be able to handle delayed offer/answer exchanges with the SIP protocol (that is, the reception of INVITEs with no Session Description Protocol).

•Route calls to Unified CME via a SIP trunk using the Unified CM dial plan configuration (route patterns, route lists, and route groups).

•Use Unified CM device pools and regions to configure a G.711 codec within the site and the G.729 codec for remote Unified CME sites.

•Configure the allow-connections sip to sip command under voice services voip on Unified CME to allow SIP-to-SIP call connections.

•For SIP endpoints, configure the mode cme command under voice register global, and configure dtmf-relay rtp-nte under the voice register pool commands for each SIP phone on Unified CME.

•For SCCP endpoints, configure the transfer-system full-consult command and the transfer-pattern .T command under telephony-service on Unified CME.

•Configure the SIP WAN interface voip dial-peers to forward or redirect calls, destined for Unified CM, with session protocol sipv2 and dtmf-relay [sip-notify | rtp-nte] on Unified CME.

via Cisco Unified Communications SRND Based on Cisco Unified Communications Manager 7.x – Call Processing   [Cisco Unified Communications Manager (CallManager)] – Cisco Systems.