Next Generation Networks [1/2] - National Taiwan …acpang/course/voip_2005/slides/...Next...
Transcript of Next Generation Networks [1/2] - National Taiwan …acpang/course/voip_2005/slides/...Next...
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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
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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
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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
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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
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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
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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.
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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
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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
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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
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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
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SIP Addressing
SIP URLs (Uniform Resource Locators)user@hostsip:[email protected]:[email protected]
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SIP Network Entities [1/4]
User AgentsA SIP-enabled telephoneUser agent client (calling party) User agent server (called party)
ServersProxy ServerRedirect ServerLocation Server (Registrar)
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Proxy ServerCan be used for call forwarding, time-of-day routing, or follow-me services
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Redirect Server
Map the destination address to zero or more new addresses
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Registrar
Accepts SIP REGISTER requestsIndicating that the user is at a particular addressPersonal/User mobility
Typically combined with a proxy or redirect server
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SIP Architecture
User Agent Client(Caller)
User Agent Server(Callee)
Proxy ServerProxy Server
Proxy Server Redirect Server Location Server
SIP RequestSIP ResponseRTP Media Stream
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SIP Call Establishment
It is simple, which contains a number of interim responses.
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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.
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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
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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
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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
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An Example of SIP RequestSIP 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: sip:[email protected];tag=c8-f3-1-4-5-3efad
User originating the request
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
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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
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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:[email protected]>To: Patric <sip:[email protected]>Call-ID: [email protected]: 1 INVITEContent-Type: applcation/sdpContent-Length: …
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Invitation for SIP Proxy Serveritri.org.tw
location server
BENZ
honda@AUDI
(5)
AUDI(1) [email protected]
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 [email protected] (8) ACK honda@AUDI
RTP Stream
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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.
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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
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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.
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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.
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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)
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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
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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
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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
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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
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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)
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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.
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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
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Network Architecture
Signaling(SS7)
Gateway
TrunkingGateway
CallAgent
SCP
ResidentialGateway
InternetSS7 Network
STP
TrunkingGatewayTrunking
GatewayCO
Switch
ResidentialGatewayResidential
Gateway
MGCP/MEGACO
MGCP/MEGACO
RTP
SIGTRAN
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MGCP Connection Establishment
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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
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The primary function of MGCP is to enableThe connections to be createdThe session descriptions to be exchanged between the connections
The Function of MGCP
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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
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Call Flow for RGW to TGW
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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
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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
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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
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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.
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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.
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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.
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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.
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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
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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
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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