WHAT IS SIP?

Session Initiation Protocol (SIP) is an application layer IETF standard protocol for initiating interactive user sessions that involve multimedia elements such as video, voice, chant, and others. SIP can establish multimedia sessions, and modify, or terminate them. SIP is based on the Web protocol HTTP, and like HTTP, SIP is a text-based, request-response protocol, utilizing TCP and/or UDP as the transport mechanism for the session messages. The SIP protocol itself is only responsible for establishing a session between two end points of a conversation, utilizing several other protocols to establish the communication services over that connection. Session Description Protocol (SDP), for example, describes the media content of the session, including the audio, video and data capabilities of the endpoint, media ports to be used for connections as well as any other information that would be pertinent to the media connections between the end points How does SIP work? SIP Signaling User dials a number (e.g. 321-103-4567) SIP-URI (Uniform Resource Identifier) is retrieved from DNS (Domain Name System). A SIP invite along with SDP (Session description protocol) formatted into an internet message format and encapsulated into Ethernet and sent via the LAN switch to the router and encapsulated into IP and UDP and SIP invite sent via TCP, UDP and other protocol to the destination proxy. The caller receives a 100 (Trying) response which indicates that the invite has been received and proxy is working to route the invite to the destination. The caller receives a 180 (ringing) response and begins ear ringing using an audio ring back tone or by displaying the message on the Telephone screen. When the person called pickups the handset, the SIP phone sends a 200 response (OK) to the caller indicating that the call has been answered. The 200 message contains a SDP media description of the type of session that the other party is willing to establish. An AKC message is sent.

SIP Media Session User begins talking. Voice is transcoded into a CODEC. (e.g. G.711. G.729) based on SDP. Voice is packetized. RTP is added and encapsulated in Ethernet and sent via the LAN Switch to the router and encapsulated into IP or IP-MPLS and UDP. If on-net IP network, the digital data is converted to an optical data stream and sent via optical fiber to an internet or internal router. If off-net, the digital data is channelized using a MG (Media Gateway (News - Alert)) to a TDM (Time Division Multiplexed) channel and sent via an optical data stream to a Class 5 CO (Central Office) switch for connections to the PSTN (Public Switched Telephone Network). User hangs up and a BYE message is sent. A 200 ACK is received confirming disconnect. User Agent A User Agent (UA) is an application that interfaces between the user and the SIP network. User Agents initiate and terminate sessions by exchanging requests and responses. User Agents operate in two fashions, but also may function as both. When

send SIP messages, the UA acts as a User Agent Client (UAC), and when receiving messages, it acts as a User Agent Server (UAS).

User Agent Client (UAC) A User Agent Client (UAC) is an application that initiates SIP requests to a User Agent Server (UAS). A UAC can be a program or a device that interacts with a user. The UAC determines the information essential for the request; the protocol, the port, and IP address of the UAS to which the request is being sent. User Agent Server (UAS) The User Agent Server (UAS) is a server application that accepts the request from a UAC and generates accept, reject, or redirect responses on behalf of the user. SIP Messages SIP is a text based protocol; and has well defined messages that are used for communications. There are two types of messages. A SIP message can be either a request from a UAC to an UAS, or a response from a UAS to a UAC. SIP Requests There are several types (methods) of SIP Requests. The most commonly used are listed below INVITE Indicates a client is being invited to participate in a session (RFC 3261) Confirms that the client has received a final response to s request (RFC 3261) Terminates a session (RFC 3261) Cancels any pending searches but does not terminate any sessions accepted (RFC 3261) Queries the capabilities of servers (RFC 3261) Registers the address listed in the To header field with a SIP server (RFC 3261) Provisional Acknowledgement (RFC 3262) Subscribes for an event of

ACK

BYE CANCEL

OPTIONS REGISTER

PRACK SUBSCRIBE

Notification from the Notifier (RFC 3265) NOTIFY PUBLISH INFO Notify the subscriber of a new Event (RFC 3265) Publishes a Event to the server (RFC 3903) Sends mid-session information that does not modify message state (RFC 2976) Ask recipient to issue SIP request (call transfer) (RFC 3515) The MESSAGE is used to transport instant messages using SIP (RFC 3428) The UPDATE method is used to modify the state of a session with out changing the state of the dialog (RFC 3311)

REFER MESSAGE

UPDATE

SIP Responses SIP Response messages contain numeric response codes. The SIP response code set is partly based on HTTP response codes. There are six classes of response: 1xx Informational Responses 100 Trying: Extended search being performed Ringing Call Is Being Forwarded Queued Session Progress

180 181 182 183 2xx Successful Responses 200 202 Ok Accepte d

3xx Redirection Responses 300 301 302 305 380 Multiple Choices Moved Permanently Moved Temporarily Use Proxy Alternative Service

4xx Client Failure Responses 400 401 404 405 406 407 Bad Request Unauthorized Not Found Method Not Allowed Not Acceptable Proxy Authentication Required Request Timeout Conflict Conditional Request Failed Request Entity Too Large Request URO Too Long Unsupported Media Type Bad Extension Temporarily Unavailable Call/Transaction

408 409 412 413 414 415 420 480 481

Does Not Exist 482 483 484 485 486 487 488 Loop Detected Too Many Hops Address Incomplete Ambiguous Busy Here Request Terminated Not Acceptable Here

5xx Server Failure Responses 500 501 Server Internal Error Not Implemented: The SIP request method is not implemented Bad Gateway Service Unavailable Server Time-out SIP Version Not Supported Message Too Large Precondition Failure

502 503 504 505 513 580

Jitter and Latency Latency can be defined as the time between a node sending a message and receipt of the message by another node. The TANDBERG systems can handle any value of latency, however, the higher the latency, the longer the delay in video and audio. This may lead to conferences with undesirable delays causing participants to interrupt and speak over each other.

Jitter can be defined as the difference in latency. Where constant latency simply produces delays in audio and video, jitter can have a more adverse effect. Jitter can cause packets to arrive out of order or at the wrong times. TANDBERG systems can manage packets with jitter up to 100ms; packets not received within this timeframe will be considered lost packets.