Use of SIP requires the following:

• SIP Service Account

  An account or accounts with a SIP internet service provider (ITSP). The method of operation and the information provided will vary. The key requirement is a SIP URI, a web address of the form name@example.com. This is the equivalent of a SIP telephone number for making and receiving calls via SIP.

• Voice Compression Channels

  SIP calls use system voice compression channels in the same way as used for standard IP trunks and extensions. For an IP500 V2 system, these are provided by the installation of VCM modules within the control unit. RTP relay is applied to SIP calls where applicable.

• Licensing

  SIP trunks require licenses in the system configuration. These set the maximum number of simultaneous SIP calls supported by the system.

• Firewall Traversal

  Routing traditional H.323 VoIP calls through firewalls often fails due to the effects of NAT (Network Address Translation). For SIP a number of ways to ensure successful firewall traversal can be used. The system does not apply any firewall between LAN1 and LAN2 to SIP calls.

  • STUN (Simple Traverse of UDP NAT)

    UDP SIP can use a mechanism called STUN to cross firewalls between the switch and the ITSP. This requires the ITSP to provide the IP address of their STUN server and the system to then select from various STUN methods how to connect to that server. The system can attempt to auto-detect the required settings to successfully connect. To use STUN, the line must be linked to the Network Topology settings of a LAN interface using the line's Use Network Topology Info setting.

  • TURN (Traversal Using Relay NAT)

    TCP SIP can use a mechanism called TURN (Traversal Using Relay NAT). This is not currently supported.

  • Session Border Control

    STUN does not have to be used for NAT traversal when SBC is between IP Office and the ITSP, since the SBCE will be performing NAT traversal.

• SIP Trunks

  These trunks are manually added to the system configuration. Typically a SIP trunk is required for each SIP ITSP being used. The configuration provides methods for multiple URI's from that ITSP to use the same trunk. For each trunk at least one SIP URI entry is required, up to 150 SIP URI's are supported on the same trunk. Amongst other things this sets the incoming and outgoing groups for call routing.

• Outgoing Call Routing

  The initial routing uses any standard short code with a dial feature. The short code's Line Group ID should be set to match the Outgoing Group ID of the SIP URI channels to use. However the short code must also change the number dialed into a destination SIP URI suitable for routing by the ITSP. In most cases, if the destination is a public telephone network number, a URI of the form 123456789@example.com is suitable. For example:

  • Code: 9N#

  • Feature: Dial

  • Telephone Number: N"@example.com"

  • Line Group ID: 100

  While this can be done in the short code, it is not an absolute necessity. The ITSP Proxy Address or ITSP Domain Name will be used as the host/domain part.

• Incoming Call Routing

  Incoming SIP calls are routed in the same way as other incoming external calls. The caller and called information in the SIP call header can be used to match Incoming CLI and Incoming Number settings in normal system Incoming Call Route records.  

• DiffServ Marking

  DiffServ marking is applied to calls using the DiffServ Settings on the System > LAN > VoIP tab of the LAN interface as set by the line's Use Network Topology Info setting.