Session Initiation Protocol (SIP) is an IP telephony signaling protocol developed by the IETF. Primarily used for Voice over Internet Protocol (VoIP) calls, SIP can also be used for video or any media type.
SIP is a text-based protocol that is based on HTTP and MIME, which makes it suitable and very flexible for integrated voice-data applications. SIP is designed for real time transmission, uses fewer resources and is considerably less complex than H.323. Its addressing scheme uses URLs and is human readable; for example: sip:john.doe@company.com.
SIP relies on the Session Description Protocol (SDP) for session description and the Real-time Transport Protocol (RTP) for actual transport.
You can configure the Experience Portal SIP connection as either a TCP connection or a TLS connection. The Avaya Aura®Session Manager (ASM) is used as the SIP proxy when configuring Experience Portal to use SIP. For more information on SIP integration with Experience Portal and ASM, see Application Notes for Avaya Experience Portal, Avaya Aura® Communication Manager, Avaya Aura® Session Manager and Acme Packet Net-Net 6.2.0 with AT&T IP Toll Free Service using MIS/PNT or AVPN Transport – Issue 1.0 on the Avaya online support Web site, http://support.avaya.com.
Note:
While configuring the SIP proxy for a system with Experience Portal zones, you must take the following into consideration:
In a system with zones, the applications, media servers, and other resources are grouped into zones.
A SIP proxy is configured for each zone.
When the target application resides in only one zone, configure the SIP proxy for the application with only those media servers that are assigned to the target zone.
If the same SIP proxy is used across multiple zones and the application is configured in multiple zones, the media servers from the multiple zones can be targeted.
