Archive for the ‘4.05 SIP’ Category

WTF is OOD Refer ?

Posted by Andy on May 6, 2009

According go SIP System Administration Guide:

Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application. The application using OOD-R triggers a call setup request that specifies the Referee address in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to communicate with Cisco Unified SRST is independent of the end-user device protocol, which can be H.323, POTS, SCCP, or SIP. Click-to-dial is an example of an application that can be created using OOD-R.

A click-to-dial application enables users to combine multiple steps into one click for a call setup. For example, a user can click a web-based directory application from his or her PC to look up a telephone number, off-hook the desktop phone, and dial the called number. The application initiates the call setup without the user having to out-dial from his or her own phone. The directory application sends a REFER message to Cisco Unified SRST, which sets up the call between both parties based on this REFER.

Figure 3 shows an example of OOD-R being used by a click-to-dial application. In this scenario, the following events occur (refer to the event numbers in the illustration):

1. Remote user clicks to dial.

2. Application sends out-of-dialog REFER to Cisco Unified SRST.

3. Cisco Unified SRST 1 connects to SIP phone 1 (Referee).

4. Cisco Unified SRST 1 sends INVITE to SRST 2.

5. Cisco Unified SRST 2 sends INVITE to SIP phone 2 (Refer-Target) and the call is accepted.

6. Voice path is created between the two SIP phones.

Note The connectivity to Cisco Unified Communications Manager has been lost and, therefore, IP phone 1 and IP phone 2 have registered to SRST routers.

Figure 3 Click-to-Dial Application using Out-of-Dialog REFER

via Cisco Unified SIP SRST System Administrator Guide – 4.1 Features  [Cisco Unified Survivable Remote Site Telephony] – Cisco Systems.


Posted in 2.02 CUCM SIP Endpoints, 4.05 SIP | Leave a Comment »

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.

Posted in 4.05 SIP | Leave a Comment »

SIP Convergency speed-up

Posted by Andy on April 15, 2009

By default the Cisco IOS SIP gateway transmits the SIP INVITE request 6 times to the Unified CM IP address configured under the dial-peer. If the SIP gateway does not receive a response from that Unified CM, it will try to contact the Unified CM configured under the other dial-peer with a higher preference.

Cisco IOS SIP gateways wait for the SIP 100 response to an INVITE for a period of 500 ms. By default, it can take up to 3 seconds for the Cisco IOS SIP gateway to reach the backup Unified CM. You can change the SIP INVITE retry attempts under the sip-ua configuration by using the command retry invite . You can also change the period that the Cisco IOS SIP gateway waits for a SIP 100 response to a SIP INVITE request by using the command timers trying under the sip-ua configuration.

One other way to speed up the failover to the backup Unified CM is to configure the command monitor probe icmp-ping under the dial-peer statement. If Unified CM does not respond to an Internet Control Message Protocol (ICMP) echo message (ping), the dial-peer will be shut down. This command is useful only when the Unified CM is not reachable. ICMP echo messages are sent every 10 seconds.

The following commands enable you to configure Unified CM redundancy on a Cisco IOS SIP gateway:


retry invite

timers trying

dial-peer voice 101 voip

destination-pattern 2…

session target ipv4:

preference 0

monitor probe icmp-ping

session protocol sipv2

dial-peer voice 102 voip

destination-pattern 2…

session target ipv4:

preference 1

monitor probe icmp-ping

session protocol sipv2

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

Posted in 4.05 SIP | Leave a Comment »