signaling protocol that controls the initiation, modification, and termination of interactive multimedia sessions
Session Initiation Protocol (SIP)
call routing and session management functions are distributed across all the nodes (including endpoints
and network servers)
Session Initiation Protocol (SIP)
In June 2002, the IETF published a new SIP RFC
provides the capability to discover the location of the end user for the purpose of establishing a session or delivering a SIP request
enables the determination of the media capabilities of the devices that are involved in the session.
willingness of the end user to engage in communication.
enables the establishment of session parameters for the parties who are involved in the session.
enables the modification, transfer, and termination of an active session.
is responsible for forwarding SIP requests to the target UAS
provides routing infromation
enforce security policy
for reserving network resources
Resource Reservation Protocol (RSVP)
for gateway selection and load balancing
Telephony Routing over IP (TRIP)
for hostname-to-IP address resolution
Domain Name System (DNS)
for preventing eavesdropping, tampering, or message forgery.
Transport Layer Security (TLS)
to resolve host or domain names into routable IP addresses
used to load-share across multiple servers in a cluster identified by a hostname
to describe the parameters of the multimedia session. This information includes session type such as audio, video, or both and parameters such as codecs or ports needed to establish a media stream.
Session Description Protocol (SDP)
transports real-time data such as audio or video packets to the endpoints that are involved in a session
Realtime Transport Protocol (RTP)
encryption technology can cause what
to reserve network resources such as bandwidth prior to establishment of the media session. This ensures that the network resources are in place prior to the called party
to provide privacy and integrity of SIP signaling information over the network
protocol to discover the presence and type of Network Address Translation (NAT) between them and the
signaling is independent of the type of session being established
can send calls to the pstn
in sip network the telephone picking up the call is the
sip default port
is typically an e-mail-type address with a format such as one of the following: sip:user@domain:port sip:user@host:port
identify a user or a resource within a network domain
are messages that are sent from client to server to invoke a SIP operation
indicates that the recipient user or service is invited to participate in a session
"placing a call"
that the UAC has received the final response to an INVITE request.
to query a UAS about its capabilities
**request the termination of a previously established session**
*hang up a call be befor the user picked up *
email address be corsponed to an ip address
sent from the server to client
1xx provisional status
numbered from 100 to 699
SIP Response Messages
establishment of the session
defined as a peer-to-peer
SIP relationship between two or more UAs that persists for the duration of the session
uses TCP, Stream Control Transmission Protocol (SCTP) and UDP
SIP transport layer
the SIP RFC 3261 encrypt the signaling information what 2
my drop out or stay in the call while it take place
SIP proxy server
proxy sends a message back to the user say that the user has moved
starts at the server not the phone
only involved in call set up
can initiate new SIP calls and modify and terminate existing calls.
creation of two distinct dialogs, which enable it to modify one SIP session without affecting the other
act as a third-party call controller (3PCC) and can establish calls between two user agents.
B2BUA(Back-to-back user agent)
To determine the address of SIP servers that support UDP transport in company.com domain, you need to
query a What ? with the DNS server
to determine what services are supported in a domain
Naming Authority Pointer (NAPTR)
the services and protocols that are supported within the domain.
Service Record (SRV)
registrar and proxy are functions implemented with the
sip proxies router SIP request to the server and sip resoponces to the
client insists that the server must understnd the SIP extension to process a request
indication to the UAS(server) that the UAC(client) understands a certain extension
SIP extension that enables a caller to express his preferences about request handling at the intermediate
Caller and Callee Preferences
which SIP nodes can request notification from remote peers when monitored events take place
(voice mail messafe changing online status)
SUBSCRIBE and NOTIFY
is waiting for the end-user input to determine whether to accept a subscription from this subscriber.
Subscription-State value is pending in the NOTIFY request
near real time mesafes but the person must be willing to engage in the session
Instant Message (IM)
enables users to publish their availability status and display messages or icons as a form of self-expression
MESSAGE requests carry the message content in the form of
an IETF signaling protocol for multimedia applications involving one or more participants
a flexible protocol that supports extensions for new applications and services
SIP dialog state is maintained at the
SIP network servers are either stateless or maintain transaction state information for at least