TSIN02 - Internetworking

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1 Lecture 8: SIP and H323 Litterature: 2004 Image Coding Group, Linköpings Universitet

2 Lecture 8: SIP and H323 Goals: After this lecture you should Understand the basics of SIP and it's architecture Understand H.323 and how it compares to SIP Understand MGCP (MEGACO/H.248) 2

3 Lecture 8: SIP and H323 Outline: Introduction Voice over IP SIP H.323 MEGACO/H.248 3

4 Introduction Voice over IP Telephony services can now be provided over IP networks. An IP telephony system needs: Signaling protocols that can locate users, set up, modify and tear down calls. SIP, H.323 Media transport protocols for transmission of packetised audio/video. RTP, TCP and UDP Supporting protocols to provide QoS, security etc. DNS, TRIP, RSVP, DIAMETER... 4

5 SIP Session Initiation Protocol - An application layer control protocol that can establish, modify, and terminate multimedia sessions (conferences) such as Internet telephony. - IETF RFC3261 5

6 SIP - Facets User location: Users can access application features from remote locations User availability: Willingness of called party to communicate User capabilities: Media and parameters to be used Session setup: Point- to- point and multiparty calls Session management: Transfer and termination, modifying session parameters, and invoking services 6

7 SIP - Features Based on HTTP- like request/response transaction model Client request invokes function on server At least one response Uses most HTTP header fields, encoding rules, and status codes Readable format for displaying information Uses concepts similar to recursive and iterative searches of DNS Incorporates Session Description Protocol (SDP) Defines session content using types similar to MIME 7

8 SIP Transport Layer SIP typically runs on UDP for performance Own reliability mechanisms May also use TCP May use Transport Layer Security (TLS) protocol for secure connection 8

9 SIP - functionality SIP is a component that can be used with other IETF protocols to build a complete multimedia architecture. RTP transports real time data and provides QoS feedback RTSP controls delivery of streaming media. MEGACO controls gateways to the Public Switched Telephone Network (PSTN). SDP describes multimedia sessions 9

10 SIP Components SIP Components Location Server Redirect Server Registrar Server PSTN User Agent Proxy Server Proxy Server Gateway 10

11 User Agents Clients sends SIP requests (initiates a call) and receives responses. Servers receives SIP requests and sends responses Both servers and clients can terminate calls 11

12 Proxy Server Acts as both a server and a client. Responds to requests directly or passes them on to other servers. Interprets, rewrites or translates a request before forwarding it. 12

13 Location Server Provides information about a called party's possible location(s) 13

14 Redirect Server Used when a user cannot be found at his/her normal address. Returns zero or more new addresses to the client. Does not initiate its own SIP requests. Does not accept or terminate calls. 14

15 Registrar Server Accepts register requests and uses the received information to update data at a location server. May support authentication Typically co- located with a proxy or redirect server and may offer location services. 15

16 SIP Messages SIP components communicate by exchanging SIP messages: SIP Methods: INVITE Initiates a call by inviting user to participate in session. ACK - Confirms that the client has received a final response to an INVITE request. BYE - Indicates termination of the call. CANCEL - Cancels a pending request. REGISTER Registers the user agent. OPTIONS Used to query the capabilities of a server. INFO Used to carry out- ofbound information, such as DTMF digits. SIP Responses: 1xx - Informational Messages. 2xx - Successful Responses. 3xx - Redirection Responses. 4xx - Request Failure Responses. 5xx - Server Failure Responses. 6xx - Global Failures Responses. 16

17 SIP Headers SIP borrows much of the syntax and semantics from HTTP. A SIP messages looks like an HTTP message message formatting, header and MIME support. An example SIP header: SIP Header INVITE sip:5120@ SIP/2.0 Via: SIP/2.0/UDP :5060 From: sip:5121@ To: <sip:5120@ > Call-ID: c e0563-2a1ce-2e323931@ CSeq: 100 INVITE Expires: 180 User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled Accept: application/sdp Contact: sip:5121@ :5060 Content-Type: application/sdp 17

18 SIP Addressing The SIP address is identified by a SIP URL, in the format: user@host. Examples of SIP URLs: sip:adam@student.liu.se sip:bob@ sip: @hotsip.com 18

19 Communication Establishement Establishing communication using SIP usually occurs in six steps: 1. Registering, initiating and locating the user. 2. Determine the media to use involves delivering a description of the session that the user is invited to. 3. Determine the willingness of the called party to communicate the called party must send a response message to indicate willingness to communicate accept or reject. 4. Call setup. 5. Call modification or handling example, call transfer (optional). 6. Call termination. 19

Registration Each time a user turns on the SIP user client (SIP IP Phone, PC, or other SIP device), the client registers with the proxy/registration server.

20 Registration Each time a user turns on the SIP user client (SIP IP Phone, PC, or other SIP device), the client registers with the proxy/registration server. Registration can also occur when the SIP user client needs to inform the proxy/registration server of its location. The registration information is periodically refreshed and each user client must re- register with the proxy/registration server. Typically the proxy/registration server will forward this information to be saved in the location/redirect server. SIP Phone User REGISTER 200 Proxy/ Registration Server REGISTER 200 SIP Messages: REGISTER Registers the address listed in the To header field. 200 OK. Location/ Redirect Server 20

21 Example 21

22 Design Framework SIP was designed for: Integration with existing IETF protocols. Scalability and simplicity. Mobility. Easy feature and service creation. 22

23 Features SIP can support these features and applications: Basic call features (call waiting, call forwarding, call blocking etc.). Unified messaging. Call forking. Click to talk. Presence. Instant messaging. Find me / Follow me. 23

24 H.323 Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real- time audio, video and/or data communications. ITU- T Recommendation H.323 Version 4 24

25 H.323 Framework H.323 defines: Call establishment and teardown. Audio visual or multimedia conferencing. 25

26 H.323 Overview H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service. 26

27 H.323 Components Gatekeeper Multipoint Control Unit Terminal Packet Based Networks Gateway Circuit Switched Networks 27

28 Terminals H.323 terminals are client endpoints that must support: H.225 call control signaling. H.245 control channel signaling. RTP/RTCP protocols for media packets. Audio codecs. Video codecs support is optional. 28

29 Gateway A gateway provides translation: For example, a gateway can provide translation between entities in a packet switched network (example, IP network) and circuit switched network (example, PSTN network). Gateways can also provide transmission formats translation, communication procedures translation, H.323 and non- H.323 endpoints translations or codec translation. 29

30 Gatekeepers Gatekeepers provide these functions: Address translation. Admission control. Bandwidth control. Zone management. Call control signaling (optional). Call authorization (optional). Bandwidth management (optional). Call management (optional). Gatekeepers are optional but if present in a H.323 system, all H.323 endpoints must register with the gatekeeper and receive permission before making a call. 30

31 Multipoint Control Unit MCU provide support for conferences of three or more endpoints. An MCU consist of: Multipoint Controller (MC) provides control functions. Multipoint Processor (MP) receives and processes audio, video and/or data streams. 31

32 Umbrella specification Media H.261 and H.263 Video codecs. H.323 G.711, G.723, G.729 Audio codecs. RTP/RTCP Media. Media Data/Fax Call Control and Signaling Data/Fax T.120 Data conferencing. T.38 Fax. Call Control and Signaling Audio Codec G.711 G.723 G.729 Video Codec H.261 H.263 RTCP T.120 T.38 H.225 Q.931 H.225 RAS H.245 H Capabilities advertisement, media channel establishment, and conference control. RTP H.225 UDP TCP TCP UDP TCP Q call signaling and call setup. RAS - registration and other admission control with a gatekeeper. IP 32

33 Communication Establishment Establishing communication in H.323 is done in five steps: 1. Call setup. 2. Initial communication and capabilities exchange. 3. Audio/video communication establishment. 4. Call services. 5. Call termination. 33

34 Comparing SIP and H323 Functionally, SIP and H.323 are similar. Both SIP and H.323 provide: Call control, call setup and teardown. Basic call features such as call waiting, call hold, call transfer, call forwarding, call return, call identification, or call park. Capabilities exchange. 34

35 Comparison different strengths H.323 Defines sophisticated multimedia conferencing. H.323 multimedia conferencing can support applications such as whiteboarding, data collaboration, or video conferencing. SIP Supports flexible and intuitive feature creation with SIP using SIP- CGI (SIP- Common Gateway Interface) and CPL (Call Processing Language). SIP Third party call control is currently only available in SIP. Work is in progress to add this functionality to H

36 MGCP Call agent or media gateway controller Provides call signaling, control and processing intelligence to the gateway. Call Agent or Media Gateway Controller (MGC) SIP H.323 Call Agent or Media Gateway Controller (MGC) Sends and receives commands to/from the gateway. MGCP MGCP Gateway Provides translations between circuit switched networks and packet switched networks. Media Gateway (MG) Media Gateway (MG) Sends notification to the call agent about endpoint events. Execute commands from the call agents. 36

37 MGCP Characteristics MGCP: A master/slave protocol. Assumes limited intelligence at the edge (endpoints) and intelligence at the core (call agent). Used between call agents and media gateways. Differs from SIP and H.323 which are peer- topeer protocols. Interoperates with SIP and H

38 MEGACO/H.248 A protocol that is evolving from MGCP and developed jointly by ITU and IETF: Megaco - IETF. H.248 or H.GCP - ITU. 38

39 Summary SIP and H.323 are comparable protocols that provide call setup, call teardown, call control, capabilities exchange, and supplementary features. MGCP is a protocol for controlling media gateways from call agents. In a VoIP system, MGCP can be used with SIP or H.323. SIP or H.323 will provide the call control functionality and MGCP can be used to manage media establishment in media gateways. 39

Real-time Services BUPT/QMUL

Real-time Services BUPT/QMUL 2017-05-27 Agenda Real-time services over Internet Real-time transport protocols RTP (Real-time Transport Protocol) RTCP (RTP Control Protocol) Multimedia signaling protocols