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
!