session initiation protocol - SIP

Editor's Notes

• #4: *Focused on but not limited to Multimedia -Voice, Video etc. *Written in clear text, human readable form -Easy to understand and implement􀂄It is a Control Protocol not a media transport protocol􀂄Distributed architecture
• #5: -One of the most active working groups in the IETF-80 active Internet drafts related to SIP
• #6: 􀂄It is a Control Protocol not a media transport protocolSIP FunctionsSIP is an enabler-protocol for VoIP and Telephony in general, due to the following features it has:Name Translation and User Location: SIP translates an address to a name and thus reaches the called party at any location. It does a mapping of session description to location, and ensures support for details of the nature of the call.Feature negotiation: Not all communicating parties (which may be more than two) have the necessary features. For example, not everyone may have video support. SIP allows the group the negotiate for the features.Call participant management: SIP allows a participant to make or cancel connections to other users during a call. Users can also be transferred or placed on hold.Call feature changes: SIP allows a user to change a call’s characteristics during the call. For example, as a user, you may want to enable of disable video, especially while a new user joins a session.
• #7: How can SIP be used? 􀂄IP Telephony –Voice & Video􀂄Instant Messaging􀂄Music and video on demand􀂄Interactive gaming􀂄Calendars, meetings, conf calls􀂄Full service web Conferencing􀂄Location based services
• #9: SIP employs design elements developed for earlier protocols. SIP is based on an HTTP-like request/response transaction model. Each transaction consists of a client request that invokes a particular method, or function, on the server and at least one response. SIP uses most of the header fields, encoding rules, and status codes of HTTP. This provides a readable text-based format for displaying information. SIP also uses concepts similar to the recursive and iterative searches of DNS. SIP incorporates the use of a Session Description Protocol (SDP), which defines session content using a set of types similar to those used in MIME.
• #11: An SIP network can be viewed of consisting of components defined on two dimensions: client/server and individual network elements. A client is any network element that sends SIP requests and receives SIP responses. A server is a network element that receives requests in order to service them and sends back responses to those requests. Stallings DCC8e Figure 24.7 shows how some of the SIP components relate to one another and the protocols that are employed. These include:• User Agent: Resides in every SIP end station. • Redirect Server: Used during session initiation to determine the address of the called device, which it returns to the calling device, directing the UAC to contact an alternate URI. This is analogous to iterative searches in DNS.• Proxy Server: An intermediary entity that acts as both a server and a client for the purpose of making requests on behalf of other clients. A proxy server primarily ensures that a request is sent to another entity closer to the targeted user. Proxies are also useful for enforcing policy. A proxy interprets, and, if necessary, rewrites specific parts of a request message before forwarding it. • Registrar: accepts REGISTER requests and places the information it receives in those requests into the location service for the domain it handles.
• #12: SIP forking refers to the process of "forking" a single SIP call to multiple SIP endpoints. This is a very powerful feature of SIP. A single call can ring many endpoints at the same time.User Agents– Can act as client and as server• Servers:– Redirect Servers• Send back alternative location of the user (similar as HTTP servers)– Proxy servers• Act on behalf of client (forwards requests)• Forking proxies– Registrars• Accepts registrations
• #14: A resource within a SIP network is identified by a Uniform Resource Identifier (URI). Examples of communications resources include the following: A user of an online service; An appearance on a multiline phone; A mailbox on a messaging system; A telephone number at a gateway service; A group (such as "sales" or "helpdesk") in an organization.SIP URIs have a format based on email address formats, namely user@domain. There are two common schemes. An ordinary SIP URI is of the form:sip:[email protected] URI may also include a password, port number, and related parameters. If secure transmission is required, "sip:" is replaced by "sips:". In the latter case, SIP messages are transported over TLS.
• #15: As was mentioned, SIP is a text-based protocol with a syntax similar to that of HTTP. There are two different types of SIP messages, requests and responses. The format difference between the two types of messages is seen in the first line. The first line of a request has a method, defining the nature of the request and a Request-URI, indicating where the request should be sent. The first line of a response has a response code. All messages include a header, consisting of a number of lines, each line beginning with a header label. A message can also contain a body, such as an SDP media description.
• #16: SIP Request – sent from client to server - is considered a 'verb‘ --RFC3261 defines six methods Description INVITEinitiate a session between two participants INVITE Requests a sessionACK Final response to the INVITEOPTIONS Ask for server capabilitiesCANCEL Cancels a pending requestBYE Terminates a sessionREGISTER Sends user’s address to server
• #17: For example, the header of message (1) in Stallings DCC8e Figure 24.11 (previous slide) might look as shown. The first line contains the method name (INVITE), a SIP URI, and the version number of SIP that is used. The Via headers show the path the request has taken in the SIP network and is used to route responses back along the same path. It contains the IP address (12.26.17.91), port number (5060), and transport protocol (UDP) that Alice wants Bob to use in his response. Subsequent proxies add additional headers.Max-Forwardslimist the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one by each proxy that forwards the request. If it reaches 0 before the request reaches its destination, it will be rejected with a 483 (Too Many Hops) error response.To contains a display name (Bob) and a SIP/SIPS URI toward which the request was originally directed. From also contains a display name (Alice) and a SIP/SIPS URI that indicate the originator of the request. It also has a tag with a random string added to the URI by the UAC to identify the session.Call-ID contains a globally unique identifier for this call, generated by the combination of a random string and the host name or IP address. CSeq or Command Sequence contains an integer and a method name. The CSeq is used to distinguish a retransmission from a new request.The Contact header contains a SIP URI for direct comms between UAs. The Content-Type indicates the type of the message body. Content-Length gives the length in octets of the message body.
• #18: Response: sent from server to a client, and provides the status of that request.SIP responses are classified by first digitStatus Codes􀂄1XX –information messages (100 –trying, 180 –ringing, 183 –progress) 􀂄2XX –successful request completion (200 –OK) 􀂄3XX –call forwarding, redirection (302 –temporarily moved, 305 –use proxy) 􀂄4XX –client error, request failure (403 –forbidden) 􀂄5XX –server error (500 –Server Internal Error, 501 –not implemented) 􀂄6XX –global failure, busy, refused, not available (606 –Not Acceptable)
• #19: For example, the header of message (11) in Stallings DCC8e Figure 24.11 (previous slide) might look as shown.The first line contains the version number of SIP that is used and the response code and name. The lines that follow are a list of header fields. The Via, To, From, Call-ID, and CSeq header fields are copied from the INVITE request. (There are three Via header field values - one added by Alice's SIP UAC, one added by the atlanta.com proxy, and one added by the biloxi.com proxy.) Bob's SIP phone has added a tag parameter to the To header field. This tag will be incorporated by both endpoints into the dialog and will be included in all future requests and responses in this call.
• #20: SIP header fields are used to convey message attributes and to modify messagemeaning. They are similar in syntax and semantics to HTTP header fields (infact some headers are borrowed from HTTP) and thus always take the format:Headers can span multiple lines. Some SIP headers such as Via, Contact, Routeand Request-Route can appear multiple times in a message or, alternatively, cantake multiple comma-separated values in a single header occurrence
• #21: START-LINE– SIP version used– In requests: address andmethod used– In responses: status code• HEADERS– Information about call• BODY (payload)– Usually SDP messageViacontains the address at which the originator is expecting to receive responses to this request. MandatoryTo contains a display name and a SIP URI towards which the request was originally directed. MandatoryDisplay names are described in RFC 2822From also contains a display name and a SIP URI that indicate the originator of the request. The From also contains a tag parameter which is used for identification purposes. MandatoryCall-ID contains a globally unique identifier for this call. MandatoryCSeq or Command Sequence contains an integer and a method name. The CSeq number is incremented for each new request within a dialog and is a traditional sequence number. MandatoryContact contains a SIP URI that represents a direct route to the originator usually composed of a username at a fully qualified domain name (FQDN). While an FQDN is preferred, many end systems do not have registered domain names, so IP addresses are permitted. The Contact header field tells other elements where to send future requests.Max-Forwards serves to limit the number of hops a request can make on the way to its destination. It consists of an integer that is decremented by one at each hop. Content-Type contains a description of the message body. MandatoryContent-Length contains an octet (byte) count of the message body.
• #27: VIA header indicates path takenby the request so farMax-Forwards field must be used with anySIP method• It limits the number for proxies or gatewayson the way of SIP message to the destination.Content-Type and Content-Lengthheader fields• Content-Type: It describes the media type ofthe message body• Content-Length: The number of octets in themessage body– It is mandatory in all SIP messages.
• #29: The call flow for SIP sessions depends upon whether the SIP session is established directly between SIP user agents or whether a SIP server (proxy, registrar, or redirect) is located between SIP user agents.Session SetupAlice's UA sends an INVITE message to Bob's SIP address (i.e., 'sip:[email protected]'). The message content is a Session Description Protocol message describing the expected media exchange.Bob's UA receives the INVITE and responds with a 100 Trying message.The UA then attempts to attract the attention of Bob, and simultaneously sends a 180 Ringing message to Alice.Bob respond and his UA sends a 200 OK message. The 200 OK contains the SDP message Bob is agreeing to.Finally, Alice's UA acknowledges receipt of the OK with an ACK request.Media streams are established directly between Alice and Bob.
• #30: The proxy server is a communication midpoint, functioning as both a user server and as a user agent. When acting as a user server the proxy receives the SIP requests and forwards them on to the destination user agent and vice versa when acting as a user agent.