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Voip Technology Overview

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Voip Technology Overview VoIP Technology Overview Ai-Chun Pang Grad. Ins. of Networking and Multimedia Dept. of Comp. Sci. and Info. Engr. National Taiwan University Outline RTP ( Real-Time Transport Protocol)/RTCP ( RT P Control Protocol) SIP ( Session Initiation Protocol) MGCP ( Media Gateway Control Protocol)/MEGACO (Me dia Ga teway Co ntrol Protocol) SIGTRAN ( Sig naling Tran sport) Softswitch 2 Voice over UDP, not TCP Speech Small packets, 10 – 40 ms Occasional packet loss is not a catastrophe. Delay-sensitive TCP: connection set-up, ack, retransmit → delays 5 % packet loss is acceptable if evenly spaced Resource management and reservation techniques (bandwidth and buffer size) A managed IP network Advanced voice-coding techniques In-sequence delivery UDP was not designed for voice traffic 3 Real-Time Transport Protocol RTP: A Transport Protocol for Real-Time Applications RFC 1889 RTP – Real-Time Transport Protocol UDP Packets may be lost or out-of-sequence RTP over UDP A sequence number A time stamp for synchronized play-out Does not solve the QoS problems; simply provides additional information 4 RTP Control Protocol RTCP A companion protocol with RTP Exchange messages between session users Quality feedback Number of lost packets, delay, inter-arrival jitter… RTCP is implicitly open when an RTP session is open E.g., RTP/RTCP uses UDP port 5004/5005 Timing of RTCP packets The control traffic should be limited to a small fraction of the session bandwidth. 5 Timing of RTCP Packets The control traffic should be limited to a small fraction of the session bandwidth. RFC 1889 provides an algorithm for calculating the interval between RTCP Packets. The following main characteristics are included. The interval > 5 seconds 0.5 – 1.5 times the calculated interval A dynamic adaptation for the interval based on the RTCP packet size 6 Introduction to SIP A powerful alternative to H.323 More flexible, simpler Easier to implement advanced features Better suited to the support of intelligent user devices A part of IETF multimedia data and control architecture 7 The Popularity of SIP Originally Developed in the MMUSIC (Multiparty Multimedia Session Control) A separate SIP working group RFC 2543 Many developers The latest version: RFC 3261 SIP + MGCP/MEGACO The VoIP signaling in the future “bake-off” Various vendors come together and test their products against each other to ensure that they have implemented the specification correctly to ensure compatibility with other implementations 8 SIP Architecture A signaling protocol The setup, modification, and tear-down of multimedia sessions SIP + SDP (Session Description Protocol) Describe the session characteristics Separate signaling and media streams 9 SIP Addressing SIP URLs (Uniform Resource Locators) user@host sip:[email protected] sip:[email protected] 10 SIP Network Entities [1/4] User Agents A SIP-enabled telephone User agent client (calling party) User agent server (called party) Servers Proxy Server Redirect Server Location Server (Registrar) 11 Proxy Server Can be used for call forwarding, time-of-day routing, or follow-me services 12 Redirect Server Map the destination address to zero or more new addresses 13 Registrar Accepts SIP REGISTER requests Indicating that the user is at a particular address Personal/User mobility Typically combined with a proxy or redirect server 14 SIP Architecture SIP Request SIP Response RTP Media Stream Proxy Server Redirect Server Location Server Proxy Server Proxy Server User Agent Client(Caller) User Agent Server(Callee) 15 SIP Call Establishment It is simple, which contains a number of interim responses. 16 Overview of SIP Messaging Syntax Text-based Similar to HTTP Disadvantage – more bandwidth consumption SIP messages message = start-line *message-header CRLF [message-body] start-line = request-line | status-line Request-line specifies the type of request The response line indicates the success or failure of a given request. 17 Overview of SIP Messaging Syntax Message headers Additional information of the request or response E.g., The originator and recipient Retry-after header Subject header Message body Describe the type of session The most common structure for the message body is SDP (Session Description Protocol). Could include an ISDN User Part message Examined only at the two ends 18 SIP Requests [1/2] Method SP Request-URI SP SIP-version CRLF Request-URI The address of the destination Methods INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER INVITE Initiate a session Information of the calling and called parties The type of media ～IAM (initial address message) of ISUP ACK only when receiving the final response 19 SIP Requests [2/2] BYE Terminate a session Can be issued by either the calling or called party OPTIONS Query capabilities A particular type of supported media CANCEL Terminate a pending request E.g., an INVITE did not receive a final response REGISTER Log in and register the address with a SIP server “all SIP servers” – multicast address (224.0.1.175) Can register with multiple servers Can have several registrations with one server 20 An Example of SIP Request SIP Request Message Description INVITE sip:[email protected] SIP/2.0 Method type, request URI and SIP version Call-ID:[email protected] Globally unique ID for this call Content-Type:application/sdp The body type, an SDP message CSeq: 1 INVITE Command Sequence number and type From: User originating the request sip:[email protected];tag=c8-f3-1-4-5-3efad To:sip:[email protected] User being invited into the call Via: SIP/2.0/UDP 140.96.200.1:8080 IP Address and port of previous hop Blank line separates header from body v=0 SDP Version o=smayer 280932498 IN IP4 140.96.200.1 Owner/creator and session identifier s=Incoming phone call from acer The name of the session p=+886 3 5914494 Phone number of caller c=IN IP4 140.96.102.100 Connection information m=audio 492837 RTP/AVP 0 Media name and transport address 21 SIP Responses SIP Version SP Status Code SP Reason-Phrase CRLF Reason-Phrase A textual description of the outcome Could be presented to the user status code A three-digit number 1XX Informational 2XX Success (only code 200 is defined) 3XX Redirection 4XX Request Failure 5XX Server Failure 6XX Global Failure All responses, except for 1XX, are considered as final responses Should be ACKed 22 An Example of SIP Response SIP/2.0 200 OK Via : SIP/2.0/UDP sippo.example.se Via : SIP/2.0/UDP science.fiction.com From : Fingal <sip:[email protected]> To : Patric <sip:[email protected]> Call-ID : [email protected] Cseq : 1 INVITE Content-Type : applcation/sdp Content-Length : … 23 Invitation for SIP Proxy Server itri.org.tw location server csie.nctu.edu.tw [email protected] honda (2) (4) INVITE (1) INVITE honda@AUDI (3) AUDI [email protected] honda@AUDI (6) 200 OK (5) 200 OK BMW BENZ (5) (7) ACK [email protected] (8) ACK honda@AUDI honda@AUDI RTP Stream 24 The Telephone Network [1/2] SS7 Signaling Service Service + ISUP Messages Control Data INAP/TCAP Messages Point Point Signal Transfer Control Layer Point Intelligent Transport Layer Peripheral Class 4 Class 5 Tandem Switch End Office Switch Circuit Switched Network 25 Reference: CCL/ITRI The Telephone Network [2/2] 5 Basic Components in Intelligent Networks SSP/Service Switching Point switching , signaling, routing, service invocation STP/Signal Transfer Point SCP SCP SDPSDP signaling, routing TCAP messages IP SCP/Service Control Point IP STP STPSTP STP service logic execution SSP SSP SDP/Service Data Point SSP ISUP messages SSP subscriber data storage, access Voice IP/Intelligent Peripheral resources such as customized voice announcement, voice recognition, DTMF digit collection 26 Why MGCP/MEGACO Voice over IP Lower cost of network implementation Integration of voice and data applications New service features Reduced bandwidth Replacing all traditional circuit-switched networks is not feasible. VoIP and circuit-switching networks must coexist. Interoperation Seamless interworking 27 Separation of Media and Call Control [1/3] Gateways Interworking To make the VoIP network appear to the circuit switched network as a native circuit-switched system and vice versa Signaling path and media path are different in VoIP systems. Media – directly (end-to-end) Signaling – through H.323 gatekeepers (or SIP proxies) SS7, Signaling System 7 The logical separation of signaling and media 28 Separation of Media and Call Control [2/3] A network gateway has two related but separate functions. Signaling conversion The call-control entities use signaling to communicate . Media conversion A slave function (mastered by call-control entities) 29 Separation of Media and Call Control [3/3] Advantages of Separation Media conversion close to the traffic source and sink The call-handling functions is centralized. A call agent (media gateway controller - MGC) can control multiple gateways. New features can be added more quickly. The first protocol is MGCP RFC 2705, IETF To be succeeded by MEGACO/H.248 Has be included in several product developments MEGACO/H.248 IETF and ITU-T Study Group 16 RFC 3015 is now the official version. 30 MGCP A master-slave protocol Call agents (MGCs) control the operation of MGs Call-control intelligence Call-related signaling MGs Do what the CA instructs A line or trunk on circuit-switched side An RTP port on the IP side Types of Media
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