SIP
The Session Initiation Protocol (SIP) is an open and lightweight protocol that does the following:
It provides mechanisms for establishing calls between a caller and a callee over an IP network.
It allows the caller to notify the callee that it wants to start a call. It allows the participants
to agree on media encodings. It also allows participants to end calls.
It provides mechanisms for the caller to determine the current IP address of the callee. Users
do not have a single, xed IP address because they may be assigned addresses dynamically
(using DHCP) and because they may have multiple IP devices, each with a di erent IP
address.
It provides mechanisms for call management, such as adding new media streams during the
call, changing the encoding during the call, inviting new participants during the call, call
transfer, and call holding.
Setting Up a Call to a Known IP Address
To understand the essence of SIP, it is best to take a look at a concrete example. In this example,
Alice is at her PC and she wants to call Bob, who is also working at his PC. Alice s and Bob s PCs
are both equipped with SIP-based software for making and receiving phone calls. In this initial
example, we ll assume that Alice knows the IP address of Bob s PC. Figure 9.9 illustrates the
SIP call-establishment process. In Figure 9.9, we see that an SIP session begins when Alice sends
Bob an INVITE message, which resembles an HTTP request message. This INVITE message is
sent over UDP to the well-known port 5060 for SIP. (SIP messages can also be sent over TCP.)
The INVITE message includes an identi er for Bob (bob@193.64.210.89), an indication of Alice s
current IP address, an indication that Alice desires to receive audio, which is to be encoded in
format AVP 0 (PCM encoded -law) and encapsulated in RTP, and an indication that she wants
to receive the RTP packets on port 38060. After receiving Alice s INVITE message, Bob sends an
SIP response message, which resembles an HTTP response message. This response SIP message
is also sent to the SIP port 5060. Bob s response includes a 200 OK as well as an indication
of his IP address, his desired encoding and packetization for reception, and his port number to
which the audio packets should be sent. Note that in this example Alice and Bob are going to use
di erent audio-encoding mechanisms: Alice is asked to encode her audio with GSM whereas Bob
is asked to encode his audio with PCM -law.
المادة المعروضة اعلاه هي مدخل الى المحاضرة المرفوعة بواسطة استاذ(ة) المادة . وقد تبدو لك غير متكاملة . حيث يضع استاذ المادة في بعض الاحيان فقط الجزء الاول من المحاضرة من اجل الاطلاع على ما ستقوم بتحميله لاحقا . في نظام التعليم الالكتروني نوفر هذه الخدمة لكي نبقيك على اطلاع حول محتوى الملف الذي ستقوم بتحميله .
|