172.16.1.1 172.16.2.1
1. Trunk 2. Route Group 3. Route List 4. Route Pattern 5. Test Trunk
IP WAN
Before deploying Cisco CallManager into a SIP environment, test the call flow between two CallManager clusters, if there are more than one cluster. This test will validate whether the CallManagers are configured correctly.
For more information, refer to the Trunk Configuration section in the Cisco CallManager Administration Guide at
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_book 09186a00802d8eaf.html.
Summary
This topic summarizes the key points discussed in this lesson.
© 2005 Cisco Systems, Inc. All rights reserved. GWGK v1.01-21
Summary
In configuring an H.323 gateway with Cisco CallManager, the first step is to set the H.225 timer to 3 seconds. The next step is to configure the dial peers.
H.323 dial peers can be configured to define primary and backup Cisco CallManager servers with switchover to a backup server if necessary.
In toll bypass situations, if the packet network bandwidth is constrained, or unavailable, the call is routed transparently to the PSTN gateway interface (dial peer #) for transport to the remote site.
MGCP PRI backhaul terminates all of the ISDN PRI Layer 2 (Q.921) signaling functions on the MGCP gateway and packages all of the ISDN PRI Layer 3 (Q.931) signaling information into packets for transmission to Cisco CallManager through an IP tunnel.
© 2005 Cisco Systems, Inc. All rights reserved. GWGK v1.01-22
Summary (Cont.)
When both SIP and H.323 are deployed in a network, support of the two protocols on a single gateway is critical. Another integral part of dual-protocol deployment is the ability for H.323 gatekeepers and SIP proxies to interwork and share routing capabilities.
In a call-processing environment that uses SIP, use SIP trunks to configure a signaling interface with Cisco CallManager for SIP calls. SIP trunks
(or signaling interfaces) connect Cisco CallManager clusters with a SIP proxy server.
When a Cisco CallManager Express router is deployed in SIP networks, its integration with SIP is via SIP gateway trunks for the support of basic calls.
Lesson Self-Check
Use the questions here to review what you learned in this lesson. The correct answers and solutions are found in the Lesson Self-Check Answer Key.
Q1) Why is it necessary to decrease the H.225 timer on the H.323 gateway to 3 seconds?
(Source: H.323 Gateway Integration with Cisco CallManager)
Q2) List the VoIP toll bypass types. (Source: H.323 Gateway Integration with Toll Bypass)
Q3) How can an H.323 voice gateway be configured to use redundant CallManagers?
(Source: H.323 Gateway Integration with Cisco CallManager)
A) H.323 gateways can use the MGCP redundant host commands if they are configured. This configuration allows H.323 and MGCP to use the primary and backup Cisco CallManager.
B) H.323 dial peers can be configured to define primary and backup CallManager servers with switchover to a backup server if necessary.
C) Use the priority keyword when defining multiple IPV4 peers in the dial plan:
always use 1 for the primary and 2 for the backup CallManager.
D) Redundancy is configured on the CallManager where the intelligence lies, not on the gateway.
Q4) Where does MGCP PRI backhaul terminate all ISDN PRI Layer 2 (Q.921) signaling functions? (Source: MCGP Backhauling)
A) MGCP PRI backhaul terminates all of the ISDN PRI Layer 2 (Q.921) signaling functions on the MGCP gateway.
B) MGCP PRI backhaul terminates all of the ISDN PRI Layer 2 (Q.921) signaling functions on the Cisco CallManager.
C) MGCP PRI backhaul cannot terminate ISDN PRI Layer 2 (Q.921) signaling; it can only terminate ISDN PRI Layer 3 (Q.931).
D) MGCP PRI backhaul terminates all of the ISDN PRI Layer 2 (Q.921) signaling functions into QSIG packets for transmission to Cisco CallManager through an IP tunnel.
Q5) What information does MGCP PRI backhaul package for transmission to a CallManager? (Source: Source: MCGP Backhauling)
A) all of the ISDN PRI Layer 2 (Q.921) signaling information B) all of the ISDN PRI Layer 3 (Q.931) signaling information C) just the display and user-to-user information elements D) just the tunneling of redirecting number information element
Q6) What are the key SIP and H.323 integration considerations? (Source: SIP Gateway Integration with Cisco CallManager)
Q7) What is required for a Cisco CallManager to make SIP calls? (Source: SIP Gateway Integration with CallManager)
Lesson Self-Check Answer Key
Q1) This step is necessary because, by default, the H.225 timer that controls redirection to a less preferred dial peer on a no-response failure is considerably longer than the 10-second timer of the Q.931 Call Proceeding timer. By setting the H.225 timer to 3 seconds, the router attempts a connection to the primary Cisco CallManager server, and if it does not receive a response in 3 seconds, it falls back to the secondary Cisco CallManager server.
Q2) Cisco CallManager to Cisco CallManager
Cisco CallManager to Cisco CallManager Express sites
Cisco CallManager to remote sites that are part of the same cluster Cisco CallManager to remote sites under PBX control
PBX to other PBX-controlled sites
Q3) B
Q4) A
Q5) B
Q6) In deployments where both SIP and H.323 protocols are used, it is important that the calls-per-second performance of both environments is similar. Provisions for communication between the Cisco SIP proxy server and H.323 gatekeepers allow hybrid networks that include both SIP and H.323 traffic.
Q7) Cisco CallManager requires an RFC 2833 DTMF-compliant MTP software device to make SIP calls. The current standard for SIP uses in-band payload types to indicate DTMF tones, and IP telephony components such as SCCP IP Phones only support out-of-band payload types. Thus, an RFC 2833-compliant MTP device monitors for payload type and acts as a translator between in-band and out-of-band payload types.