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Title: SIP Trunking With Call Manager Express

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For many years now, telephony voice services for businesses and enterprises have been provided by using legacy PBX systems connected to the Public Switched Telephone Network (PSTN) using TDM connections (T1/E1 ISDN PRI lines or BRI or analog lines). This is shown on the figure below:



Newer telephony systems adopted the IP technology on the internal LAN, but they still used TDM connectivity (ISDN PRI/BRI and analog lines) to connect to the legacy PSTN network as shown below:



The newest trend is to go all-IP using SIP TRUNKING to connect your business office to the Telephony Service Provider network. A SIP Trunk allows the company to replace the traditional TDM fixed lines (PRI, BRI etc) with just a normal IP connection towards the service provider. This solution offers significant cost savings to the enterprise as you avoid costly BRI/PRI lines. Also, voice/data traffic can be converged on a single IP connection. This scenario is shown below:



The Cisco Call Manager Express product can be used as the telephony SIP trunk gateway between the local IP telephony network and the IP Telephony Service Provider. Calls from and to PSTN will be handled by a SIP PROXY server located in the Service Provider network.

A sample Call Manager Express configuration for SIP trunking is shown below (a snippet of the complete configuration is shown):



voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mycompany.test.com

voice class codec 1
codec preference 1 g711ulaw

!— Inbound Translation Rule
!—  for Auto Attendant pilot number “500″
voice translation-rule 1
rule 1 /5552222100/ /500/

voice translation-profile AutoAttendant
!— Applied to the inbound dial-peers for AA
translate called 1

!— SIP Trunk Configuration —
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming AutoAttendant
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
no vad

dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 9……….
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad

dial-peer voice 3 voip
description **International Outgoing Call to SIP Trunk**
destination-pattern 9011T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad

!— SIP UA Configuration —
sip-ua
authentication username 5552222100 password 075A701E1D5E415447425B
no remote-party-id
retry invite 2
retry register 10
retry options 0
timers connect 100
registrar dns: mycompany.test.com expires 3600
sip-server dns: mycompany.test.com
host-registrar
!


Source from: http://www.cisco-tips.com/sip-trunking-with-call-manager-express/

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Images: http://www.cisco-tips.com/images/legacy-pbx-pstn.jpg

http://www.cisco-tips.com/images/cme-pstn.jpg

http://www.cisco-tips.com/images/cme-sip-trunking.jpg



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Post date: 2009-04-21 11:52:58

Post date GMT: 2009-04-21 03:52:58

Post modified date: 2010-07-23 13:15:06

Post modified date GMT: 2010-07-23 05:15:06



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