User's Manual
Avaya Branch Gateway Manager 10.0 Page 237
15-601011 Issue 29r (Friday, November 02, 2012)B5800 Branch Gateway
Configuration Settings: Line
SIP URIs
Calls across SIP require URI's (Uniform Resource Identifiers), one for the source and one for the destination. Each SIP
URI consists of two parts, the user part (for example name) and the domain part (for example example.com) to form a
full URI (in this case name@example.com). SIP URI's can take several forms:
· name@117.53.22.2
· name@example.com
· 012345678@example.com
Typically each account with a SIP service provider will include a SIP URI or a set of URI's. The domain part is then used
for the SIP trunk configured for routing calls to that provider. The user part can be assigned either to an individual user if
you have one URI per user for that ITSP, or it can also be configured against the line for use by all users who have calls
routed via that line.
· If the wildcard * is used in the SIP trunk's Local URI, Contact and Display fields, that SIP trunk will accept any
incoming SIP call. The incoming call routing is still performed by the system incoming call routes based on
matching the values received with the call or the URI's incoming group setting. For outgoing calls using this SIP
URI, all valid short code CLI manipulations are used (transforming calling party number to ISDN will be ignored).
Resource Limitation
A number of limits can affect the number of SIP calls. When one of these limits is reached the following occurs: any
further outgoing SIP calls are blocked unless some alternate route is available using ARS; any incoming SIP calls are
queued until the required resource becomes available. Limiting factors are:
· the number of licensed SIP channels.
· the number of SIP channels configured for a SIP URI.
· the number of voice compression channels.
· SIP Line Call to/from Non-IP Devices
Voice compression channel required.
· Outgoing SIP Line Call from IP Device
No voice compression channel required.
· Incoming SIP Line Call to IP Device
If using the same codec, voice compression channel reserved until call connected. If using differing codecs
then 2 channels used.
SIP Information Display
The full from and to SIP URI will be recorded for use by SMDR. For all other applications and for telephone devices, the
SIP URI is put through system directory matching the same as for incoming CLI matching. First a match against the full
URI is attempted, then a match against the user part of the URI. Directory wildcards can also be used for the URI
matching.
SIP Standards
The system implementation of SIP conforms to the following SIP RFC's.
RFC
Description
2833 [7]
RTP Payload for DTMF digits, telephony tones and telephony signals.
3261 [8]
SIP Session Initiation Protocol.
3263
Locating SIP Services
3264 [11]
An Offer/Answer Model with Session Description Protocol (SDP).
3323 [14]
A Privacy Mechanism for SIP
3489 [18]
STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators
(NAT's).
3824 [24]
Using E.164 Numbers with the Session Initiation Protocol (SIP). E.164 is the ITU-T recommendation for
international public telecommunication numbering plans.