1

Next Generation Networks [1/2]

WirelessLAN WLAN

CAFASIP

ProxyServer

Internet

MS

3GPPCSCF

3G UMTS

GGSN

SGSN

T-SGW

MGW

MGW PSTN

3GPPMGCF

Internet Telecom & Wireless Communication

Reference: CCL/ITRI

2

Next Generation Networks [2/2]Internet Telecom & Wireless Communication

IP

MGCF

CSCF

T-SGW MGWMGW

WLAN

GPRS

CSCFSIP

Server

PSTN

InternetWireless App.Server

3rd Parties App.

Reference: CCL/ITRI

3

Outline

RTP (Real-Time Transport Protocol)/RTCP (RTP Control Protocol)SIP (Session Initiation Protocol)MGCP (Media Gateway Control Protocol)/MEGACO (Media Gateway Control Protocol)SIGTRAN (Signaling Transport)Softswitch

4

Voice over UDP, not TCPSpeech

Small packets, 10 – 40 msOccasional packet loss is not a catastrophe.Delay-sensitive

TCP: connection set-up, ack, retransmit → delays

5 % packet loss is acceptable if evenly spacedResource management and reservation techniques (bandwidth and buffer size)A managed IP networkAdvanced voice-coding techniques

In-sequence deliveryUDP was not designed for voice traffic

5

Real-Time Transport ProtocolRTP: A Transport Protocol for Real-Time Applications

RFC 1889RTP – Real-Time Transport Protocol

UDPPackets may be lost or out-of-sequence

RTP over UDPA sequence numberA time stamp for synchronized play-outDoes not solve the QoS problems; simply provides additional information

6

RTP Control ProtocolRTCPA companion protocol with RTPExchange messages between session usersQuality 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 5004/5005Timing of RTCP packets

The control traffic should be limited to a small fraction of the session bandwidth.

7

Timing of RTCP PacketsThe 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 seconds0.5 – 1.5 times the calculated intervalA dynamic adaptation for the interval based on the RTCP packet size

8

Introduction to SIP

A powerful alternative to H.323More flexible, simplerEasier to implement advanced featuresBetter suited to the support of intelligent user devicesA part of IETF multimedia data and control architecture

9

The Popularity of SIPOriginally Developed in the MMUSIC (Multiparty Multimedia Session Control)

A separate SIP working groupRFC 2543Many developersThe latest version: RFC 3261

SIP + MGCP/MEGACOThe 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 correctlyto ensure compatibility with other implementations

10

SIP Architecture

A signaling protocolThe setup, modification, and tear-down of multimedia sessions

SIP + SDP (Session Description Protocol)Describe the session characteristics

Separate signaling and media streams

11

SIP Addressing

SIP URLs (Uniform Resource Locators)user@hostsip:collins@home.netsip:3344556789@telco.net

12

SIP Network Entities [1/4]

User AgentsA SIP-enabled telephoneUser agent client (calling party) User agent server (called party)

ServersProxy ServerRedirect ServerLocation Server (Registrar)

13

Proxy ServerCan be used for call forwarding, time-of-day routing, or follow-me services

14

Redirect Server

Map the destination address to zero or more new addresses

15

Registrar

Accepts SIP REGISTER requestsIndicating that the user is at a particular addressPersonal/User mobility

Typically combined with a proxy or redirect server

16

SIP Architecture

User Agent Client(Caller)

User Agent Server(Callee)

Proxy ServerProxy Server

Proxy Server Redirect Server Location Server

SIP RequestSIP ResponseRTP Media Stream

17

SIP Call Establishment

It is simple, which contains a number of interim responses.

18

Overview of SIP Messaging Syntax

Text-basedSimilar to HTTPDisadvantage – more bandwidth consumption

SIP messagesmessage = start-line

*message-header CRLF[message-body]

start-line = request-line | status-lineRequest-line specifies the type of requestThe response line indicates the success or failure of a given request.

19

Overview of SIP Messaging Syntax

Message headersAdditional information of the request or responseE.g.,

The originator and recipientRetry-after headerSubject header

Message bodyDescribe the type of sessionThe most common structure for the message body is SDP (Session Description Protocol).Could include an ISDN User Part messageExamined only at the two ends

20

SIP Requests [1/2]Method SP Request-URI SP SIP-version CRLFRequest-URI

The address of the destinationMethods

INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTERINVITE

Initiate a sessionInformation of the calling and called partiesThe type of media～IAM (initial address message) of ISUPACK only when receiving the final response

21

SIP Requests [2/2]BYE

Terminate a sessionCan be issued by either the calling or called party

OPTIONSQuery capabilities

A particular type of supported media

CANCELTerminate a pending requestE.g., an INVITE did not receive a final response

REGISTERLog in and register the address with a SIP server“all SIP servers” – multicast address (224.0.1.175)Can register with multiple serversCan have several registrations with one server

22

An Example of SIP RequestSIP Request Message Description INVITE sip:acer12345689@itri.org.tw SIP/2.0 Method type, request URI and SIP version Call-ID:123456789@itri.org.tw 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: sip:1732489@itri.org.tw;tag=c8-f3-1-4-5-3efad

User originating the request

To:sip:acer12345689@itri.org.tw 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

23

SIP ResponsesSIP Version SP Status Code SP Reason-Phrase CRLFReason-Phrase

A textual description of the outcomeCould be presented to the user

status codeA three-digit number1XX Informational2XX Success (only code 200 is defined)3XX Redirection4XX Request Failure5XX Server Failure6XX Global FailureAll responses, except for 1XX, are considered as final responses

Should be ACKed

24

An Example of SIP Response

SIP/2.0 200 OKVia: SIP/2.0/UDP sippo.example.seVia: SIP/2.0/UDP science.fiction.com From: Fingal <sip:ffl@fiction.com>To: Patric <sip:pgn@example.se>Call-ID: 1234567890@science.fiction.comCseq: 1 INVITEContent-Type: applcation/sdpContent-Length: …

25

Invitation for SIP Proxy Serveritri.org.tw

location server

BENZ

honda@AUDI

(5)

AUDI(1) INVITEhonda@itri.org.tw

acer@csie.nctu.edu.tw

csie.nctu.edu.tw

BMW(2

) hon

da

(3) h

onda

@A

UD

I

(4) INVITEhonda@AUDI

(5) 200 OK(6) 200 OK

(7) ACK honda@itri.org.tw (8) ACK honda@AUDI

RTP Stream

26

SIP Extensions and Enhancements

RFC 2543, March 1999SIP has attracted enormous interest.Traditional telecommunications companies, cable TV providers and ISP

A large number of extensions to SIP have been proposed.SIP will be enhanced considerably before it becomes an Internet standard.

27

SIP INFO Method

Specified in RFC 2976For transferring information during an ongoing session

The transfer of DTMF digitsThe transfer of account balance information

Pre-paid serviceThe transfer of mid-call signaling informationA powerful, flexible tool to support new services

28

SIP for Instant Messaging

The IETF working group – SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE)A new SIP method – MESSAGE

This request carries the actual message in a message body.A MESSAGE request does not establish a SIP dialog.

29

SIP REFER Method

To enable the sender of the request to instruct the receiver to contact a third party

With the contact details for the third party included within the REFER requestFor Call Transfer applications

The Refer-to: and Refer-by: HeadersThe dialog between Mary and Joe remains established.

Joe could return to the dialog after consultation with Susan.

30

Reliability of Provisional Responses

Provisional Responses100 (trying), 180 (ringing), 183 (session in progress)Are not answered with an ACK

The messages is sent over UDPUnreliable

Lost provisional response may cause problems when interoperating with other network.

180, 183 → Q931 alerting or ISUP ACMTo drive a state machine

The new method: PRACK (Provisional Response ACK)

31

Class 5 End Office Switch

The Telephone Network [1/2]

Circuit Switched Network

Intelligent Peripheral

Signal Transfer Point

Service Control Point

Class 4 Tandem Switch

Service Data Point

+

Transport Layer

Control Layer

SS7 SignalingISUP MessagesINAP/TCAP Messages

Reference: CCL/ITRI

32

The Telephone Network [2/2]

5 Basic Components in Intelligent NetworksSSP/Service Switching Point

switching, signaling, routing, service invocation

STP/Signal Transfer Pointsignaling, routing

SCP/Service Control Pointservice logic execution

SDP/Service Data Pointsubscriber data storage, access

IP/Intelligent Peripheralresources such as customized voice announcement, voice recognition, DTMF digit collection

SSPSSP

SCPSCP SDPSDP

STPSTPIPIP

SSPSSP

STPSTP

TCAP messages

ISUP messages

Voice

33

Why MGCP/MEGACO

Voice over IPLower cost of network implementationIntegration of voice and data applicationsNew service featuresReduced bandwidth

Replacing all traditional circuit-switched networks is not feasible.VoIP and circuit-switching networks must coexist.

InteroperationSeamless interworking

34

Separation of Media and Call Control [1/3]

GatewaysInterworkingTo 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 VoIPsystems.

Media – directly (end-to-end)Signaling – through H.323 gatekeepers (or SIP proxies)

SS7, Signaling System 7The logical separation of signaling and media

35

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)

36

Separation of Media and Call Control [3/3]

Advantages of SeparationMedia conversion close to the traffic source and sinkThe 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 MGCPRFC 2705, IETFTo be succeeded by MEGACO/H.248Has be included in several product developments

MEGACO/H.248IETF and ITU-T Study Group 16RFC 3015 is now the official version.

37

MGCPA master-slave protocol

Call agents (MGCs) control the operation of MGsCall-control intelligenceCall-related signaling

MGsDo what the CA instructsA line or trunk on circuit-switched sideAn RTP port on the IP side

Types of Media GatewayTrunking Gateway to CO/SwitchesResidential Gateway to PSTN PhonesAccess Gateway

Communication between call agentsLikely to be the SIP

38

Network Architecture

Signaling(SS7)

Gateway

TrunkingGateway

CallAgent

SCP

ResidentialGateway

InternetSS7 Network

STP

TrunkingGatewayTrunking

GatewayCO

Switch

ResidentialGatewayResidential

Gateway

MGCP/MEGACO

MGCP/MEGACO

RTP

SIGTRAN

39

MGCP Connection Establishment

40

Endpoints, Connections and CallsEndpoints

Sources or sinks of mediaTrunk interfacesPOTS line interfacesAnnouncement endpoint

ConnectionsAllocation of IP resources to an endpointAn ad hoc relationship is established from a circuited-switched line and an RTP port on the IP side.A single endpoint can have several connections

A callA group of connections

41

The primary function of MGCP is to enableThe connections to be createdThe session descriptions to be exchanged between the connections

The Function of MGCP

42

9 commands to handle Connection/EndpointsEndpointConfiguration (coding characteristics)NotificationRequest (requested events)Notify (GW: detected events)CreateConnectionModifyConnectionDeleteConnectionAuditEndpointAuditConnectionRestartInProgress (GW : taken in/out of service)

All commands are acknowledged.

EPCF RQNTNTFYCRCXMDCXDLCXAUEPAUCXRSIP

MGCP Commands

43

Call Flow for RGW to TGW

46

H323 (SIP) vs. MGCP(MEGACO)

GWGK

MCU

GW : GatewayGK : GatekeeperTN : TerminalMCU : Multipoint Control Unit

TN

PSTNCA

TGW RGW

CA : Call AgentTGW : Trunking GatewayRGW : Residential GatewaySG : Singling Gateway

SS7

PSTN CO

SG

RTP

MGCP

H.323

TNTN

GWGK

MCU

TN

TNTN

Reference: CCL/ITRI

47

Relation with H.323 Standards

COSwitch

Internet

CallAgent

Signaling(SS7)

Gateway

TrunkingGateway

Gatekeeper

Terminalor

Gateway

SS7/ISUP

RTP

H.225/RASH.225/Q931H.245

SIGTRAN

MGCP

Reference: CCL/ITRI

48

Signaling TransportAddressing the issues regarding the transport of signaling within IP networks

The issues related to signaling performance within IP networks and the interworking with PSTN

Issues discussed in SIGTRANAddress translationHow can we deploy an SS7 application (e.g., ISUP) that expects certain services from lower layers such as MTP when lower layers do not exist in the IP network?For transport layer, the ISUP message must be carried in the IP network with the same speed and reliability as in the SS7.

UDP x TCP x

49

SIGTRAN ArchitectureSignaling over standard IP uses a common transport protocol that ensures reliable signaling delivery.

Stream Control Transmission Protocol (SCTP)An adaptation layer is used to support specific primitives as required by a particular signaling application.

The standard SS7 applications (e.g., ISUP) do not realize that the underlying transport is IP.

50

Why not use TCP and UDP?UDP

Unreliable transmissionLack of congestion and flow controls

TCPTCP provides both reliable data transfer and strict order-of-transmission, but SS7 may not need ordering.

TCP will cause delay for supporting order-of-transmission.Head-of-line blocking

The limited scope of TCP sockets complicates the task of data transmission using multi-homed hosts.TCP is relatively vulnerable to DoS attack, such as SYN attacks.

51

SCTPTo offer the fast transmission and reliability required for signaling carrying.SCTP provides a number of functions that are critical for telephony signaling transport.

It can potentially benefit other applications needing transport with additional performance and reliability.

SCTP must meet the Functional Requirements of SIGTRAN.

52

What Supported By Using SCTP?

To ensure reliable, error-free, in-sequence delivery of user messages (optional).To support fast delivery of messages and avoid head-of-line blocking.To support network-level fault tolerance that is critical for carrier-grade network performance by using multi-home hosts.To provide protection against DoS attack by using 4-way handshake and cookie.

53

An Example of SIGTRAN

ISUPTCAP

SCCP

MAP

MTP

OSI Layers

Application

PresentationSession

Transport

Network

Data Link

Physical

INAP

SS7 Protocol Stack

ISUPTCAP

SCCP

MAP

SCN Signaling Adaptation(SSA)

Common Signaling Transport(CST)

IP

INAP

SIGTRAN Protocol Stack

54

Softswitch Overview

Providing open layered architecture

Circuit-Switched

TransportHardware

Call Control & Switching

Services & Applications

PROPRIETARY

• Solutions in a proprietary box• Little room for innovation

Soft-Switched

Transport Hardware

Softswitch Call Control

Services, Applications & Features (Management, Provisioning and

Back Office)

Open Protocols APIs

Open Protocols APIs

• Solutions are based on open standards• Open standards enable lower cost for

innovation

Reference: CCL/ITRI

55

Softswitch Architecture

Softswitch: Emulating Circuit Switching in Software

IN/SCPPSTNLocal Switch

PSTNLocal Switch STP

SS7 Network

IP Network

RTP Streams

MGCMGC MGCMGC

Trunk Trunk GatewayGateway

Trunk Trunk GatewayGateway

SGSGSGSG

IP PhoneIP Phone

90009000 Personalized VoIPService System

Application ServerApplication Server

Reference: CCL/ITRI