User's Manual

Avaya Branch Gateway Manager 10.0 Page 236
15-601011 Issue 29r (Friday, November 02, 2012)B5800 Branch Gateway
5.4.11 SIP Line
Use of SIP requires the following:
1.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 [e-mail]. This is the
equivalent of a SIP telephone number for making and receiving calls via SIP.
2.Voice Compression Channels
SIP calls use system voice compression channels in the same way as used for standard IP trunks and extensions.
These are provided by the installation of VCM modules within the control unit. RTP relay is applied to SIP calls where
applicable.
3.Licensing
SIP trunks require licenses in the system configuration. These set the maximum number of simultaneous SIP calls
supported by the system.
4.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 is not required is the ITSP if a Session Border Controller (SBC) is used between the system and the ITSP.
The system does not perform its own SBC.
5.SIP Trunks
These trunks are manually added to the system configuration. Typically a SIP trunk is required for each SIP ITSP being
used. As 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.
6.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 [e-mail] is suitable. For example:
· Code: 9N#
· Feature: Dial
· Telephone Number: N"@example.com"
· Line Group ID: 100
7.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.
8.DiffServ Marking
DiffServ marking is applied to calls using the DiffServer Settings on the System | LAN | VoIP tab of the LAN
interface as set by the line's Use Network Topology Info setting.
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