Department of Computer Science. Burapha University 6 SIP (I)

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1 Burapha University ก Department of Computer Science 6 SIP (I) Functionalities of SIP Network elements that might be used in the SIP network Structure of Request and Response SIP messages Other important protocols for establishing SIP session Ver. 0.1 ก :, prajaks@buu.ac.th

2 SIP Introduction Functionality Network Elements SIP Message Building Blocks Addressing Messages Message Structure Interaction with Other IETF Protocols ก : Internet Telephony, 6: SIP (I) 2 ก 36

3 SIP Introduction ก : Internet Telephony, 6: SIP (I) 3 ก 36

4 SIP Introduction SIP - Session Initiation Protocol Signaling protocol that controls the initiation, modification, and termination of interactive multimedia sessions Multimedia sessions could be as diverse as audio or video calls among two or more parties, chat sessions, or game sessions Extensions -- instant messaging, presence, and event notifications text-based protocol peer-to-peer protocol -- no centralized switches defined in RFC 2543 (March 1999) by the Multiparty Multimedia Session Control (MMUSIC) -- IETF new SIP RFC (RFC 3261) June 2002 ก : Internet Telephony, 6: SIP (I) 4 ก 36

5 SIP Introduction SIP Functionality User location SIP provides the capability to discover the location of the end user for the purpose of establishing a session or delivering a SIP request. User mobility is inherently supported in SIP. User capabilities SIP enables the determination of the media capabilities of the devices that are involved in the session. User availability SIP enables the determination of the willingness of the end user to engage in communication. ก : Internet Telephony, 6: SIP (I) 5 ก 36

6 SIP Introduction SIP Functionality Session setup SIP enables the establishment of session parameters for the parties who are involved in the session. Session handling SIP enables the modification, transfer, and termination of an active session. ก : Internet Telephony, 6: SIP (I) 6 ก 36

7 SIP Introduction SIP Network Elements User agent A user agent (UA) is a logical function in the SIP network that initiates or responds to SIP transactions. A UA can act as either the client or the server in a SIP transaction. A UA might or might not directly interact with a human user. A UA is stateful that is, it maintains session or dialog state. User agent client A user agent client (UAC) is a logical function that initiates SIP requests and accepts SIP responses. Examples of UAC are a SIP phone initiating a call on behalf of a human user or a SIP Proxy forwarding a request on behalf of a UAC. User agent server A user agent server (UAS) is a logical function that accepts SIP requests and sends back SIP responses. A SIP phone accepting an INVITE request is one example. ก : Internet Telephony, 6: SIP (I) 7 ก 36

8 SIP Introduction SIP Network Elements Proxy A proxy is an intermediate entity in the SIP network that is responsible for forwarding SIP requests to the target UAS or another proxy on behalf of the UAC. A proxy primarily provides the routing function in the SIP network. A proxy might also enforce policy in the network, such as authenticating a user before providing him with service. A proxy can be stateless, transaction stateful, or call stateful. Typically, proxies are transaction stateful that is, they maintain state for the duration of a transaction (about 32 seconds). Redirect server A redirect server is a UAS that generates 300 class SIP responses to requests it receives, directing the UAC to contact an alternate set of Uniform Resource Identifiers (URI). ก : Internet Telephony, 6: SIP (I) 8 ก 36

9 SIP Introduction SIP Network Elements Registrar server A registrar is a UAS that accepts SIP REGISTER requests and updates the information from the request message into a location database. Back-to-back user agent A back-to-back user agent (B2BUA) is an intermediate entity that processes incoming SIP requests as a UAS. To answer the incoming SIP request, the B2BUA acts as a UAC, regenerates a SIP request, and sends it on the network. A B2BUA must maintain dialog state and participates in all transactions within the dialog. ก : Internet Telephony, 6: SIP (I) 9 ก 36

10 SIP Introduction SIP Message Flow ก : Internet Telephony, 6: SIP (I) 10 ก 36

11 ก : Internet Telephony, 6: SIP (I) 11 ก 36

12 SIP Addressing identifies a user or a resource within a network domain typically referred to as SIP URI typically an -type address with a format such as sip:user@domain:port sip:user@host:port user field identifies a user by name, such as john.doe, or by telephone number, such as , within the context of a domain or a host port is an optional field -- default is 5060 The public SIP address of a user or a resource is referred to as an Address-of- Record (AOR) ก : Internet Telephony, 6: SIP (I) 12 ก 36

13 SIP Addressing Examples: RFC 3261 specifies a secure SIP URI format SIPS URI sips:user@domain:port sips:user@host:port The default port for a SIPS URI is 5061 ก : Internet Telephony, 6: SIP (I) 13 ก 36

14 SIP Messages devided into SIP requests and responses SIP Requests sent from client to server to invoke a SIP operation RFC 3261 defines six SIP requests to locate users and initiate, modify, and tear down sessions INVITE ACK OPTIONS BYE CANCEL REGISTER ก : Internet Telephony, 6: SIP (I) 14 ก 36

15 SIP Requests INVITE indicates that the recipient user or service is invited to participate in a session modify the characteristics of a previously established session might include the description of the media session being set up or modified, encoded per SDP (Session Description Protocol) A successful response (200 OK response) to an INVITE indicates the willingness of the called party to participate in the resulting media session ACK request confirms that the UAC has received the final response to an INVITE request. used only with INVITE requests. sent end to end for a 200 OK response. can include a message body with the final session description if the INVITE request did not contain a session description. ก : Internet Telephony, 6: SIP (I) 15 ก 36

16 SIP Requests OPTIONS A UA uses the OPTIONS request to query a UAS about its capabilities. If the UAS is capable of delivering a session to the user, it responds with the capability set of the UAS. BYE A UA uses BYE to request the termination of a previously established session. CANCEL The CANCEL request enables UACs and network servers to cancel an in-progress request, such as INVITE. This does not affect completed requests in which the UAS had already sent final responses. REGISTER A client uses a REGISTER request to register its current location information corresponding to the AOR of the user with SIP servers. ก : Internet Telephony, 6: SIP (I) 16 ก 36

17 SIP Responses A server sends a SIP response to a client to indicate the status of a SIP request The UAS or proxy generates SIP responses in response to a SIP request that the UAC initiates SIP responses are numbered from 100 to 699. SIP responses are grouped as 1xx, 2xx, and so on through 6xx SIP responses are classified as provisional and final A provisional response indicates progress by the server -- not final outcome 1xx A final response indicates the termination and the final status of a SIP request -- 2xx, 3xx, 4xx, 5xx, and 6xx ก : Internet Telephony, 6: SIP (I) 17 ก 36

18 SIP Responses 1xx class of SIP response indicates provisional status 100 Trying 180 Ringing 181 Call is being forwarded 182 Queued 183 Session progress A 2xx class response indicates successful processing of the SIP request. 200OK ก : Internet Telephony, 6: SIP (I) 18 ก 36

19 SIP Responses A 3xx class response indicates that the SIP request needs to be redirected to another UAS for processing. 300 Multiple choices 301 Moved permanently 302 Moved temporarily 305 Use proxy 380 Alternative service ก : Internet Telephony, 6: SIP (I) 19 ก 36

20 SIP Responses A 4xx, 5xx, or 6xx class of response indicates failure in processing of the SIP request. 400 Bad request 401 Unauthorized 402 Payment required 403 Forbidden 404 Not found 500 Internal server error 600 Busy everywhere 603 Decline ก : Internet Telephony, 6: SIP (I) 20 ก 36

21 SIP Message Structure consists of the following: A start-line One or more header fields An empty line indicating the end of header fields An optional message body start-line, each message-header line, and the empty line must be terminated by a Carriage Return Line Feed (CRLF) start-line for a SIP request is a Request-Line SIP method, the Request-URI, and the SIP version start-line for a SIP response is a Status-line SIP version, the SIP response code, and an optional reason phrase ก : Internet Telephony, 6: SIP (I) 21 ก 36

22 SIP Message Structure ก : Internet Telephony, 6: SIP (I) 22 ก 36

23 SIP Message Structure SIP Request message ก : Internet Telephony, 6: SIP (I) 23 ก 36

24 SIP Message Structure SIP Request message ก : Internet Telephony, 6: SIP (I) 24 ก 36

25 SIP Message Structure SIP Response message ก : Internet Telephony, 6: SIP (I) 25 ก 36

26 SIP Message Structure SIP Response message ก : Internet Telephony, 6: SIP (I) 26 ก 36

27 SIP Message Structure SIP Headers conveys the signaling and routing information for the SIP network entities follows the same format as an HTTP header (RFC 2616) consists of a field name followed by a colon (:) and the field value From: Alice <sip:alice@company.com>;tag= To: Bob <sip:bob@proxy.company.com>;tag= ก : Internet Telephony, 6: SIP (I) 27 ก 36

28 SIP Message Structure SIP Headers From This header indicates the identity of the initiator of a SIP request. The From header is usually the AOR of the sender. It consists of a SIP or SIPS URI and an optional display name. To This header indicates the desired recipient of a SIP request. The To header is usually the AOR of the recipient. The SIP request might not always be delivered to the "desired" recipient because of redirection or forwarding. The To header consists of a SIP or SIPS URI and an optional display name. ก : Internet Telephony, 6: SIP (I) 28 ก 36

29 SIP Message Structure SIP Headers Call-ID This header field identifies a series of SIP messages. Call-ID must be identical for all SIP requests and responses sent by either UA within a dialog. Cseq This header is composed of an integer value and method-name. This header identifies, orders, and sequences SIP requests within a dialog. The Cseq header also differentiates between message retransmissions and new messages. ก : Internet Telephony, 6: SIP (I) 29 ก 36

30 SIP Message Structure SIP Headers Via The Via header indicates the path taken by the request and identifies where the response needs to be sent. Contact This header identifies a SIP or SIPS URI where the UA wants to receive a new SIP request. Allow The Allow header lists the set of SIP methods supported by the UA that is generating the message. ก : Internet Telephony, 6: SIP (I) 30 ก 36

31 SIP Message Structure SIP Headers Supported This header lists all SIP extensions supported by the UA. SIP extensions are SIP RFCs other than RFC SIP extensions are represented as option tags such as 100rel defined in RFC Require This header has similar semantics to the Supported header, but the support of the SIP extension at the remote UA is a must for the transaction to be processed. ก : Internet Telephony, 6: SIP (I) 31 ก 36

32 SIP Message Structure SIP Headers Content-Type This header indicates the type of the message body that is attached to a SIP request or response. This header must be present if the SIP message has a body. Content-Length This header indicates the size of the message body (in decimal) in a SIP message. This header is a must when SIP messages are carried over stream-based protocols such as TCP. ก : Internet Telephony, 6: SIP (I) 32 ก 36

33 Interaction with Other IETF Protocols ก : Internet Telephony, 6: SIP (I) 33 ก 36

34 Interaction with Other IETF Protocols Interaction with Other IETF Protocols DNS SIP session establishment might require the use of DNS to resolve host or domain names into routable IP addresses. DNS can also be used to loadshare across multiple servers in a cluster identified by a hostname. Session Description Protocol (SDP) SDP is used in a SIP message body 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. RFC 2327 defines SDP. ก : Internet Telephony, 6: SIP (I) 34 ก 36

35 Interaction with Other IETF Protocols Interaction with Other IETF Protocols Real-time Transport Protocol (RTP) RTP, first defined in RFC 1889, transports real-time data such as audio or video packets to the endpoints that are involved in a session. Real-time Transport Control Protocol (RTCP), defined in RFC 1890, provides quality of service (QoS) feedback to the sender. RFC 3550 obsoletes RFC RSVP (Resource Reservation Setup Protocol) SIP can use RSVP 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 being alerted about an incoming call. ก : Internet Telephony, 6: SIP (I) 35 ก 36

36 Interaction with Other IETF Protocols Interaction with Other IETF Protocols TLS (Transport Layer Security) SIP recommends the use of TLS, defined in RFC 2246, to provide privacy and integrity of SIP signaling information over the network. TLS allows the client and server applications to authenticate each other, negotiate encryption algorithms, and establish cryptographic keys before sending the signaling information over the network. STUN (Simple Traversal of UDP Through Network Address Translators) to discover the presence and type of Network Address Translation (NAT) between them and the public Internet. allows the client to discover the public IP address that is allocated to the NAT. This procedure works for most types of NAT except symmetric NAT. Symmetric NAT occurs when all requests from the same internal IP address and port to a specific destination IP address and port are mapped to the same external source IP address and port. ก : Internet Telephony, 6: SIP (I) 36 ก 36

SIP Compliance APPENDIX

SIP Compliance APPENDIX APPENDIX E This appendix describes Cisco SIP proxy server (Cisco SPS) compliance with the Internet Engineering Task Force (IETF) definition of Session Initiation Protocol (SIP) as described in the following