1
Next Generation Networks [1/2]
Wireless
LAN WLAN
FA CA
ProxySIP Server
Internet
MS
CSCF3GPP
3G UMTS
GGSN SGSN
T-SGW
MGW
MGW PSTN
MGCF3GPP z Internet Telecom & Wireless Communication
Reference: CCL/ITRI
Next Generation Networks [2/2]
z Internet Telecom & Wireless Communication
IP
MGCF CSCF
T-SGW MGW MGW
WLAN GPRS
CSCF SIP
Server
PSTN
Internet
Wireless App.
Server 3rd Parties App.
Reference: CCL/ITRI
3
Outline
z RTP (Real-Time Transport Protocol)/RTCP (RTP Control Protocol)
z SIP (Session Initiation Protocol)
z MGCP (Media Gateway Control Protocol)/MEGACO (Media Gateway Control Protocol)
z SIGTRAN (Signaling Transport)
z Softswitch
Voice over UDP, not TCP
z Speech
z Small packets, 10 – 40 ms
z Occasional packet loss is not a catastrophe.
z Delay-sensitive
z TCP: connection set-up, ack, retransmit → delays
z 5 % packet loss is acceptable if evenly spaced
z Resource management and reservation techniques (bandwidth and buffer size)
z A managed IP network
z Advanced voice-coding techniques
z In-sequence delivery
z UDP was not designed for voice traffic
5
Real-Time Transport Protocol
z RTP: A Transport Protocol for Real-Time Applications
z RFC 1889
z RTP – Real-Time Transport Protocol
z UDP
z Packets may be lost or out-of-sequence
z RTP over UDP
z A sequence number
z A time stamp for synchronized play-out
z Does not solve the QoS problems; simply provides additional information
RTP Control Protocol
z RTCP
z A companion protocol with RTP
z Exchange messages between session users
z Quality feedback
z Number of lost packets, delay, inter-arrival jitter…
z RTCP is implicitly open when an RTP session is open
z E.g., RTP/RTCP uses UDP port 5004/5005
z Timing of RTCP packets
z The control traffic should be limited to a small fraction of the session bandwidth.
7
Timing of RTCP Packets
z The control traffic should be limited to a small fraction of the session bandwidth.
z RFC 1889 provides an algorithm for calculating the interval between RTCP Packets.
z The following main characteristics are included.
z The interval > 5 seconds
z 0.5 – 1.5 times the calculated interval
z A dynamic adaptation for the interval based on the RTCP packet size
Introduction to SIP
z A powerful alternative to H.323
z More flexible, simpler
z Easier to implement advanced features
z Better suited to the support of intelligent user devices
z A part of IETF multimedia data and control architecture
9
The Popularity of SIP
z Originally Developed in the MMUSIC (Multiparty Multimedia Session Control)
z A separate SIP working group
z RFC 2543
z Many developers
z The latest version: RFC 3261
z SIP + MGCP/MEGACO
z The VoIP signaling in the future
z “bake-off”
z Various vendors come together and test their products against each other
z to ensure that they have implemented the specification correctly
z to ensure compatibility with other implementations
SIP Architecture
z A signaling protocol
z The setup, modification, and tear-down of multimedia sessions
z SIP + SDP (Session Description Protocol)
z Describe the session characteristics
z Separate signaling and media streams
11
SIP Addressing
z SIP URLs (Uniform Resource Locators)
z user@host
z sip:collins@home.net
z sip:3344556789@telco.net
SIP Network Entities [1/4]
z User Agents
z A SIP-enabled telephone
z User agent client (calling party)
z User agent server (called party)
z Servers
z Proxy Server
z Redirect Server
z Location Server (Registrar)
13
Proxy Server
z Can be used for call forwarding, time-of-day routing, or follow-me services
Redirect Server
z Map the destination address to zero or more new addresses
15
Registrar
z Accepts SIP REGISTER requests
z Indicating that the user is at a particular address
z Personal/User mobility
z Typically combined with a proxy or redirect server
SIP Architecture
User Agent Client(Caller)
User Agent Server(Callee) Proxy Server Proxy Server
Proxy Server Redirect Server Location Server SIP Request
SIP Response RTP Media Stream
17
SIP Call Establishment
z It is simple, which contains a number of interim responses.
Overview of SIP Messaging Syntax
z Text-based
z Similar to HTTP
z Disadvantage – more bandwidth consumption
z SIP messages
z message = start-line
*message-header CRLF [message-body]
z start-line = request-line | status-line
z Request-line specifies the type of request
z The response line indicates the success or failure of a given request.
19
Overview of SIP Messaging Syntax
z Message headers
z Additional information of the request or response
z E.g.,
z The originator and recipient
z Retry-after header
z Subject header z Message body
z Describe the type of session
z The most common structure for the message body is SDP (Session Description Protocol).
z Could include an ISDN User Part message
z Examined only at the two ends
SIP Requests [1/2]
z Method SP Request-URI SP SIP-version CRLF
z Request-URI
z The address of the destination
z Methods
z INVITE, ACK, OPTIONS, BYE, CANCLE, REGISTER
z INVITE
z Initiate a session
z Information of the calling and called parties
z The type of media
z ～IAM (initial address message) of ISUP
z ACK only when receiving the final response
21
SIP Requests [2/2]
z BYE
z Terminate a session
z Can be issued by either the calling or called party
z OPTIONS
z Query capabilities
z A particular type of supported media
z CANCEL
z Terminate a pending request
z E.g., an INVITE did not receive a final response
z REGISTER
z Log in and register the address with a SIP server
z “all SIP servers” – multicast address (224.0.1.175)
z Can register with multiple servers
z Can have several registrations with one server
An Example of SIP Request
SIP 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 Responses
z SIP Version SP Status Code SP Reason-Phrase CRLF
z Reason-Phrase
z A textual description of the outcome
z Could be presented to the user
z status code
z A three-digit number
z 1XX Informational
z 2XX Success (only code 200 is defined)
z 3XX Redirection
z 4XX Request Failure
z 5XX Server Failure
z 6XX Global Failure
z All responses, except for 1XX, are considered as final responses
z Should be ACKed
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:ffl@fiction.com>
To: Patric <sip:pgn@example.se>
Call-ID: 1234567890@science.fiction.com Cseq: 1 INVITE
Content-Type: applcation/sdp Content-Length: …
25
Invitation for SIP Proxy Server
itri.org.tw
location server
BENZ
honda@AUDI (5) (1) INVITE AUDI
honda@itri.org.tw acer@csie.nctu.edu.tw
csie.nctu.edu.tw
BMW
(2) honda (3) honda@AUDI (4) INVITE
honda@AUDI (5) 200 OK (6) 200 OK
(7) ACK honda@itri.org.tw (8) ACK honda@AUDI
RTP Stream
SIP Extensions and Enhancements
z RFC 2543, March 1999
z SIP has attracted enormous interest.
z Traditional telecommunications companies, cable TV providers and ISP
z A large number of extensions to SIP have been proposed.
z SIP will be enhanced considerably before it becomes an Internet standard.
27
SIP INFO Method
z Specified in RFC 2976
z For transferring information during an ongoing session
z The transfer of DTMF digits
z The transfer of account balance information
z Pre-paid service
z The transfer of mid-call signaling information
z A powerful, flexible tool to support new services
SIP for Instant Messaging
z The IETF working group – SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE)
z A new SIP method – MESSAGE
z This request carries the actual message in a message body.
z A MESSAGE request does not establish a SIP dialog.
29
SIP REFER Method
z To enable the sender of the request to instruct the receiver to contact a third party
z With the contact details for the third party included within the REFER request
z For Call Transfer applications
z The Refer-to: and Refer-by: Headers
z The dialog between Mary and Joe remains established.
z Joe could return to the dialog after consultation with Susan.
Reliability of Provisional Responses
z Provisional Responses
z 100 (trying), 180 (ringing), 183 (session in progress)
z Are not answered with an ACK
z The messages is sent over UDP
z Unreliable
z Lost provisional response may cause problems when interoperating with other network.
z 180, 183 → Q931 alerting or ISUP ACM
z To drive a state machine
z 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 Signaling
ISUP Messages
INAP/TCAP Messages
Reference: CCL/ITRI
The Telephone Network [2/2]
z 5 Basic Components in Intelligent Networks
z SSP/Service Switching Point
z switching, signaling, routing, service invocation
z STP/Signal Transfer Point
z signaling, routing
z SCP/Service Control Point
z service logic execution
z SDP/Service Data Point
z subscriber data storage, access
z IP/Intelligent Peripheral
z resources such as customized voice announcement, voice recognition, DTMF digit collection
SSPSSP
SCPSCP
SDPSDP
STPSTP IPIP
SSPSSP STPSTP
TCAP messages
ISUP messages Voice
33
Why MGCP/MEGACO
z
Voice over IP
z Lower cost of network implementation
z Integration of voice and data applications
z New service features
z Reduced bandwidth
z
Replacing all traditional circuit-switched networks is not feasible.
z
VoIP and circuit-switching networks must coexist.
z Interoperation
z Seamless interworking
Separation of Media and Call Control [1/3]
z Gateways
z Interworking
z To make the VoIP network appear to the circuit
switched network as a native circuit-switched system and vice versa
z Signaling path and media path are different in VoIP systems.
z Media – directly (end-to-end)
z Signaling – through H.323 gatekeepers (or SIP proxies)
z SS7, Signaling System 7
z The logical separation of signaling and media
35
Separation of Media and Call Control [2/3]
z A network gateway has two related but separate functions.
z Signaling conversion
z The call-control entities use signaling to communicate.
z Media conversion
z A slave function (mastered by call-control entities)
Separation of Media and Call Control [3/3]
z Advantages of Separation
z Media conversion close to the traffic source and sink
z The call-handling functions is centralized.
z A call agent (media gateway controller - MGC) can control multiple gateways.
z New features can be added more quickly.
z The first protocol is MGCP
z RFC 2705, IETF
z To be succeeded by MEGACO/H.248
z Has be included in several product developments
z MEGACO/H.248
z IETF and ITU-T Study Group 16
z RFC 3015 is now the official version.
37
MGCP
z A master-slave protocol
z Call agents (MGCs) control the operation of MGs
z Call-control intelligence
z Call-related signaling
z MGs
z Do what the CA instructs
z A line or trunk on circuit-switched side
z An RTP port on the IP side
z Types of Media Gateway
z Trunking Gateway to CO/Switches
z Residential Gateway to PSTN Phones
z Access Gateway
z Communication between call agents
z Likely to be the SIP
Network Architecture
Signaling (SS7) Gateway
Trunking Gateway
Call Agent SCP
Residential Gateway
Internet
SS7 Network
STP
Trunking GatewayTrunking
Gateway CO
Switch
Residential GatewayResidential
Gateway MGCP/
MEGACO
MGCP/
MEGACO
RTP
SIGTRAN
39
MGCP Connection Establishment
Endpoints, Connections and Calls
z Endpoints
z Sources or sinks of media
z Trunk interfaces
z POTS line interfaces
z Announcement endpoint
z Connections
z Allocation of IP resources to an endpoint
z An ad hoc relationship is established from a circuited- switched line and an RTP port on the IP side.
z A single endpoint can have several connections
z A call
z A group of connections
41
z The primary function of MGCP is to enable
z The connections to be created
z The session descriptions to be exchanged between the connections
The Function of MGCP
z 9 commands to handle Connection/Endpoints
z EndpointConfiguration (coding characteristics)
z NotificationRequest (requested events)
z Notify (GW: detected events)
z CreateConnection
z ModifyConnection
z DeleteConnection
z AuditEndpoint
z AuditConnection
z RestartInProgress (GW : taken in/out of service)
z All commands are acknowledged.
EPCF RQNT NTFY CRCX MDCX DLCX AUEP AUCX RSIP
MGCP Commands
43
Call Flow for RGW to TGW
46
H323 (SIP) vs. MGCP(MEGACO)
GK GW
MCU
GW : Gateway GK : Gatekeeper TN : Terminal
MCU : Multipoint Control Unit TN
PSTN
CA
TGW RGW
CA : Call Agent
TGW : Trunking Gateway RGW : Residential Gateway SG : Singling Gateway
SS7
PSTN CO
SG
RTP MGCP
H.323 TN
TN
GW GK
MCU
TN
TN TN
47
Relation with H.323 Standards
CO Switch
Internet
Call Agent Signaling
(SS7) Gateway
Trunking Gateway
Gatekeeper
Terminal or Gateway SS7/ISUP
RTP
H.225/RAS H.225/Q931 H.245
SIGTRAN
MGCP
Reference: CCL/ITRI
Signaling Transport
z Addressing the issues regarding the transport of signaling within IP networks
z The issues related to signaling performance within IP networks and the interworking with PSTN
z Issues discussed in SIGTRAN
z Address translation
z How 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?
z For transport layer, the ISUP message must be carried in the IP network with the same speed and reliability as in the SS7.
z UDP x
z TCP x
49
SIGTRAN Architecture
z Signaling over standard IP uses a common transport protocol that ensures reliable signaling delivery.
z Stream Control Transmission Protocol (SCTP)
z An adaptation layer is used to support specific primitives as required by a particular signaling application.
z The standard SS7 applications (e.g., ISUP) do not realize that the underlying transport is IP.
Why not use TCP and UDP?
z UDP
z Unreliable transmission
z Lack of congestion and flow controls
z TCP
z TCP provides both reliable data transfer and strict order- of-transmission, but SS7 may not need ordering.
z TCP will cause delay for supporting order-of-transmission.
z Head-of-line blocking
z The limited scope of TCP sockets complicates the task of data transmission using multi-homed hosts.
z TCP is relatively vulnerable to DoS attack, such as SYN attacks.
51
SCTP
z To offer the fast transmission and reliability required for signaling carrying.
z SCTP provides a number of functions that are critical for telephony signaling transport.
z It can potentially benefit other applications needing transport with additional performance and reliability.
z SCTP must meet the Functional Requirements of SIGTRAN.
What Supported By Using SCTP?
z To ensure reliable, error-free, in-sequence delivery of user messages (optional).
z To support fast delivery of messages and avoid head-of- line blocking.
z To support network-level fault tolerance that is critical for carrier-grade network performance by using multi-home hosts.
z To provide protection against DoS attack by using 4-way handshake and cookie.
53
An Example of SIGTRAN
TCAP ISUP SCCP
MAP
MTP
OSI Layers Application Presentation
Session Transport
Network Data Link
Physical
INAP
SS7 Protocol Stack
TCAP ISUP SCCP
MAP
SCN Signaling Adaptation
(SSA)
Common Signaling Transport
(CST)
IP INAP
SIGTRAN Protocol Stack
Softswitch Overview
z Providing open layered architecture
Circuit-Switched
Transport Hardware Call Control
& Switching Services &
Applications P
R O P R I E T A R Y
• 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
z Softswitch: Emulating Circuit Switching in Software
IN/SCP
PSTN
Local Switch PSTN
Local Switch STP
SS7 Network
IP Network
RTP Streams
MGCMGC MGCMGC
Trunk Trunk Gateway
Gateway Trunk Trunk
Gateway Gateway SGSG
SG SG
IP Phone IP Phone
9000
9000 Personalized VoIP Service System
Application Server Application Server
Reference: CCL/ITRI
