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Internet Telephony: VoIP, SIP

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Internet Telephony: VoIP, SIP & more Shivkumar Kalyanaraman: shiv rpi Adapted from s of Henning Schulzrinne, Doug Moeller Telephony: history and evolution ... – PowerPoint PPT presentation

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Title: Internet Telephony: VoIP, SIP


1
Internet Telephony VoIP, SIP more
  • Shivkumar Kalyanaraman

shiv rpi
Adapted from slides of Henning Schulzrinne, Doug
Moeller
2
Overview
  • Telephony history and evolution
  • IP Telephony What, Why Where?
  • Adding interactive multimedia to the web
  • Being able to do telephony on IP with a variety
    of devices
  • Consumer business markets
  • Key element of convergence in carrier
    infrastructure
  • Basic IP telephony model
  • Protocols SIP, H.323, RTP, Coding schemes,
    Megaco
  • Future Invisible IP telephony and control of
    appliances

3
What is VoIP? Why VoIP?Where is VoIP Today?
4
What is VoIP?
  • VoIP Voice over IP
  • Transmission of telephony services via IP
    infrastructure
  • gt need history/concepts reg. both telephony
    (or voice) and IP
  • Complements or replaces other Voice-over-data
    architecture
  • Voice-over-TDM
  • Voice-over-Frame-Relay
  • Voice-over-ATM
  • First proprietary IP Telephony implementations in
    1994, VoIP-related standards available 1996
  • Buzzwords related to VoIP
  • H.323 v2, SIP, MEGACO/H.248, Sigtrans

5
What is VoIP? Protocol Soup
SDP
MGCP
SGCP
SIP
H.323
Megaco
IPDC
MDCP
The nice thing about standards is that you have
so many to choose from furthermore, if you do
not like any of them, you can just wait for next
years model. Tanenbaum
Q.SIG
Sigtrans
H.GCP
VPIM
H.245
6
Telephony over IP standards bodies
  • ITU - International Telecommunication Union
  • http//www.itu.org
  • IETF - Internet Engineering Task Force.
  • http//www.ietf.org
  • ETSI - European Telecommunications Standards
    Institute
  • http//www.etsi.org/tiphon
  • ANSI - American National Standards Institute
  • http//www.ansi.org
  • TIA - Telecommunications Industry Association
  • http//www.tiaonline.org
  • IEEE - Institute for Electrical and Electronics
    Engineers
  • http//www.ieee.org

7
Why VoIP? Telephony Mature Industry
ATT Divestiture
8
Why VoIP Price/call plummeting due to
overcapacity
ATT Divestiture
1996 deregulation
9
Relevant Telecom Industry Trends
  • 1984 ATT breakup baby bells vs long distance
    carriers
  • 1996 Telecom deregulation, Internet takeoff
  • Late 1990s explosion of fiber capacity in
    long-distance many new carriers
  • Long-distance prices plummet
  • Despite internet, the last-mile capacity did not
    grow fast enough
  • 2000s shakeout consolidation in developed
    countries
  • Wireless substitution in last mile gt cell phone
    instead of land-lines
  • Developing countries leap frog to cell phones
  • 3G, WiMax gt broadband, VoIP mobility
  • Broadband rollouts happening slowly, but picking
    up steam now.
  • Cable offering converged bundled services
  • digital cable, VoIP, video
  • Recent mergers ATT (long-distance data
    network provider) bought by SBC (baby bell)
    Verizon/Qwest vs MCI saga

10
Why VoIP ? Data vs Voice Traffic
Note quantity ? quality ? value-added Intera
ctive svcs (phone, cell, sms) still dominate on
a -per-Mbps basis
Infrastructure convergence Since we are building
future networks for data, can we slowly junk the
voice infrastructure and move over to IP?
11
Trends Total Phone vs Data Revenues
12
Motivations and drivers
Class-4/5 switches bulky, expensive. Incentive to
switch to cheaper easily managed IP
PSTN
Class 4 switch
Class 5 switch
Voice
Class 5 switch
Users
Users
ISDN Switch
H.323 gateway
Data
Initial gateway between PSTN and Internet was
H.323. Gateway did signaling, call control,
translation in one box. Not scalable.
Packet networks
13
Voice Over IP Marketplace Drivers
  • Rate arbitrage declining but still has importance
    as cost driver
  • TDM origination and termination with IP transport
    in the WAN
  • International settlement and domestic access cost
    avoidance
  • Enterprises seeking to save on intra-company
    calls and faxes on converged network
  • Emergence of native IP origination environments
  • IP PBX, IP Phones, Soft Phones, Multimedia on the
    LAN
  • 3G Wireless, Broadband Networks
  • Companies web-based call centers/web
    callback/e-commerce with IP Enablement
  • New network-based IP features and services
  • Hosted IP PBX/IP Centrex , Unified Messaging,
    Multimedia Conferencing
  • Presence Mobility, Follow me, Teleworker, Voice
    Portal Services, WiFi
  • Technology maturing with open standards for
    easier, faster innovation
  • Converging Local, long-distance (LD) and data
    services

14
VoIP Volumes Are Accelerating While Adoption of
Applications is Growing
M of Minutes
VoIP VPN Traffic
Enterprise Adoption of VoIP / IPT Applications
Respondents
M ofMinutes
Virtual PBX Managed IP PBX traffic
Source Giga Group, "Next Generation IP Telephony
Applications Deliver Strategic Business Value",
October 20, 2003
  • VoIP VPNs will continue to be driven by increased
    IPT deployments in larger enterprises, coupled
    with economic benefits accruing, especially for
    MNCs
  • IPT Deployments are the leading edge market
    driver for the development of converged LANs and
    WANs

Source Probe Research Inc. Reaching the Big
Guys Global Enterprise Forecast. September
2002
15
Drivers Are Evolving From Cost Savings to Added
Business Value
  • Cost Savings
  • Toll By-Pass
  • Effective Use of Bandwidth
  • Personnel / Staffing Efficiencies
  • Less Expensive Moves, Adds Changes
  • Convergence / Consolidation
  • Decreased Capital
  • Upgrading to an IP PBX
  • Increased Investment Protection
  • Contact Center Functions
  • Future Proofing Infrastructure
  • Leveraging embedded infrastructure with a phased
    roll-out
  • Networking Expertise for Integration From Concept
    to Deployment
  • Optimized Business Value
  • Services over IP
  • Consistent Client / User Experience
  • Integrated Infrastructure

16
Summary Why VoIP?
  • Cost reduction
  • Toll by-pass
  • WAN Cost Reduction
  • Lowered Infrastructure Costs
  • Operational Improvement
  • Simplification of Routing Administration
  • LAN/Campus Integration
  • Policy and Directory Consolidation
  • Business Tool Integration
  • Voice mail, email and fax mail integration
  • Mobility enabled by IP networking
  • Web Overseas Call Centers
  • Collaborative applications
  • New Integrated Applications

3Cs Convergence Costs Competition
17
Where is VoIP? Consumer VoIP Markets
  • Convergence Competition
  • Vonage pure VoIP CLEC (300K subscribers)
  • Cable companies
  • Eg Time Warner (220K subscribers and signing on
    10K per week (end of 2004))
  • Bundled with digital cable services
  • Skype (computer-computer p2p VoIP) tens of
    millions
  • Also has a WiFi service a product co-developed
    by Motorola (over 3G networks)
  • Long-distance providers ATT CallVantage
  • Local (ILECs) Verizon
  • Future convergence of VoIP WiMax (802.16) as a
    open low-cost competitor to 3G wireless (closed
    system)
  • Combines broadband Internet, mobility and VoIP

18
Consumer VoIP over broadband
Broadband Infrastructure
Residential Media Gateway
Media Gateway Controller
Traditional phone
Signaling and media gateways To reach PSTN or
other networks
19
Consumer VoIP at home with cable
PacketCable standard with DOCSIS 1.1 access
infrastructure
Call Management Server
Media Gateway
Cable Modem Term. Sys.
MGC
Signaling Gateway
Cable Modem
MTA (Media Terminal Adapter)
Other access mechanisms will similarly hand over
to an MGC
20
Consumer VoIP ATT CallVantage
  • New consumer services
  • Personal conferencing earlier available to
    businesses only
  • Prepaid Calling cards offering personal
    conferencing
  • Portable TA (terminal adaptor) can plug into any
    ethernet jack or WiFi (eg many hotels providing
    free internet)
  • Universal messaging voice messages in email
  • LocateMe,
  • Do-Not-Disturb,
  • Unified Portal

21
Skype p2p VoIP over Internet
  • Skype is entirely peer-to-peer and is equivalent
    to two H.323 terminals or 2 SIP terminals talking
    to each other
  • Provides a namespace
  • Efficient coding of voice packets
  • Instant messaging with voice
  • Uses Kazaa-like p2p directory secure
    authentication (login server) and e2e encryption

22
VoIP over Wireless
  • Cellular networks with 2.5G and 3G have packet
    services
  • 1xRTT on 2.5 G
  • EV-DO on 3G
  • The voice on these networks is circuit switched
    voice
  • However,
  • Combined with bluetooth or USB interfaces, a
    PC-based VoIP software can do VoIP anywhere there
    is cellular coverage.
  • Or Cellphone can be a SIP terminal
  • Near Future VoIP over WiMax (802.16) and WiFi
    (802.11) networks

23
Enterprise Private Branch Exchange (PBX)
Post-divestiture phenomenon...
7040
212-8538080
External line
7041
Telephone switch
Corporate/Campus
Private Branch Exchange
Another switch
7042
7043
Internet
Corporate/Campus LAN
24
Enterprise VoIP Yesterdays networks
Circuit Switched Networks (Voice)
PBX
CO
PBX
CO
CO
Headquarters
Branch Offices
Router
Router
Router
Router
Router
Packet Switched Networks (IP)
25
Enterprise VoIP Todays networksToll by-pass
Circuit Switched Networks (Voice)
PBX
PBX
CO
CO
CO
Headquarters
Branch Offices
Router
Router
Router
Router
Router
Packet Switched Networks (IP)
26
Enterprise VoIP Tomorrows networksUnified/Conve
rged Networks
CO
CO
Legacy PSTN
Router
Router
Router
Router
Router
Unified Networks (Voice over IP)
Headquarters
Branch Offices
27
ATTs Integrated Infrastructure Supports
Multiple Endpoints, Access Technologies and
Application Services
  • VoIP infrastructure is converged onto a single
    IP/ MPLS network
  • Open standards architecture based on SIP protocol
  • Call Control Element manages all SIP signaling
    within our core network
  • Access Agnostic TDM, ATM, Frame, MIS, IP
    Enabled Frame and EVPN
  • Border Elements translate the multiple
    protocols into SIP, provide compression and
    security
  • Provides secure, integrated voice / data / video
    access
  • Flexibility to support future applications

Voice Applications IP Centrex, IP Call Center
and Distant Worker
ATT Call Control Element
Common VoIP Connectivity Layer
NG Border Elements
SIP Border Elements
H.323 Border Elements
MGCP Border Elements
IP/MPLS Converged Network
28
VoIP Network UtilitiesEnsure Seamless Operations
  • Outbound Call
  • IP to Circuit Switched

Circuit Switched Network
  • Inbound Call
  • Circuit Switched to IP

Network Adjunct
Customer Records
  • 800 Call
  • Circuit Switched to IP

Media Server
App. Server
App. Server
Gateway
Softswitch
  • Redirect Call
  • Circuit Switched to IP

IP Network
  • SDN Call
  • IP to Circuit Switched

29
IP-enabled circuit switches
  • PBX with VoIP trunk card
  • trunk between PBX
  • Key system or PBX with VoIP line card
  • for IP phones

VoIP Gateway
CO
Switch
30
Telephony-enabled packet networks
  • Enterprise Router with telco interfaces
  • T1/PRI
  • BRI
  • Branch office router with telco interfaces
  • BRI
  • Analog trunk/line
  • Analog dongle
  • a few analog lines for fax/phone

Central Office
Router
VoIP Gateway
31
VoFR (Voice over Frame Relay)
  • FRF.11 standard
  • Allows for G.711, 729, 728, 726, and 723.1
  • Signaling is done by transporting CAS natively or
    CCS as data
  • Has support for T.30 Fax, and Dialed Digits
    natively

Router
Switch
PBX
VFRAD
VFRAD
PBX
Switch
Switch
32
Voice over Packet Market Forecast North
America
33
Telephony History, Review Trends
34
VoIP Where Does it Fit in Trends ?
  • Phase 1 Analog Networks
  • Voice carried as analog signal
  • Phase 2 Digital Networks the rise of the
    Internet
  • Network is digital analog conversion at end
    systems
  • Benefits Noise ?, capacity?
  • Egs TDM and T-hierarchy (T1, T3, SONET etc)
  • Used as the base for the internet private data
    networks
  • Phase 3 Voice-over-X
  • Voice over Packets VoFR, VoIP
  • Key Voice moves to a higher layer (from layer 1)
  • I.e. an app over a frame relay, ATM or IP network
  • VoIP Sales pitch Convergence, Choice, Services,
    Integration with Web applications
  • Better chance of convergence compared to earlier
    attempts ISDN, B-ISDN

35
Public Telephony (PSTN) History
  • 1876 invention of telephone
  • 1915 first transcontinental telephone (NYSF)
  • 1920s first automatic switches
  • 1956 TAT-1 transatlantic cable (35 lines)
  • 1962 digital transmission (T1)
  • 1965 1ESS analog switch
  • 1974 Internet packet voice
  • 1977 4ESS digital switch
  • 1980s Signaling System 7 (out-of-band)
  • 1990s Advanced Intelligent Network (AIN)

36
PSTN Evolution
Office Switched W/ Hierarchy
Full Mesh
Office Switched
37
ATT Telephony Hierarchy
Class 1
Class 2
Class 3
Class 4
Class 5
Source Computer Networks, Andrew S. Tanenbaum
38
PSTN early days 40s-60s
  1. In-band signaling voice and control channel same
  2. Complex and dedicated hardware
  3. Hard to add new apps like caller-id, 800 calling
    etc

Tandem Office
Local Office
Local Office
User A
User B
39
Advanced Intelligent Network
  • Out-of-band signaling
  • Introduce adv services like caller-id easily
  • Reduced wastage of circuits in voice network
  • Signaling could be over a packet network
  • E.g. SS7 stack

Signaling Network
Customer Info for Advanced services
Voice Network
Local Office
User A
User B
Sometimes also called Intelligent Network,
arrival of services other than voice
40
The PSTN Architecture
  • PSTN Public Switched Telephone Network
  • Uses digital trunks between Central Office
    switches (CO)
  • Uses analog line from phones to CO

Analog
Digital
Analog
41
The PSTN Digitization
  • Voice frequency is 100 - 5000 Hz, with the main
    portion from 300 3400 Hz
  • Nyquist Theorem states that sampling must be done
    at twice the highest frequency to recreate. 4000
    Hz was chosen as the maximum frequency, thus
    sampling at 8000 Hz
  • PCM 8kHz 8 bits per sample 64 kbit/s

42
Quantization
43
Companding
44
The PSTN Digitization
  • The PCM encoding used in the PSTN is standardized
    as G.711 by the ITU
  • Each sample is represented by one byte
  • The voice signal is companded to improve voice
    quality at low amplitude levels (Which most
    conversation is at)
  • The ITU standards for companding are called A-law
    and u-law
  • G.711 A-law is used in Europe
  • G.711 ?-law is used in the US and Japan

45
The PSTN Digital Voice Transmission
  • The digital trunks between the COs are based upon
    the T-carrier system, developed in the 1960s
  • Each frame carries one sample (8 bits) for each
    24 channels, plus one framing bit 193 bits
  • 193 8000 (samples/sec) 1.544 Mbit/sec T-1

Channel 1
Channel 2
Framing Bit
Channel 3
Channel 1
Channel 2
Channel 3
Channel 24

TDM

Channel 24
1 D4 Frame
46
The PSTN Architecture, Switches
  • PSTN Public Switched Telephone Network
  • As the name says, its switched
  • Each conversation requires a channel switched
    throughout the network
  • Circuit setup uses a separate out-of-band
    intelligent network (SS7)

1. Call is requested
3. Channel is established
2. Call is accepted
47
Legacy Digital Circuit Switch
  • Centralized Intelligence
  • Proprietary Code
  • Proprietary service deployment
  • Very expensive

48
Whats the difference between a Class 5 and a
Class 4 switch?
  • Class 5
  • Located at the edge of the network
  • Trunk to Line/Line to Line
  • Aprox. 30,000 deployed
  • Services Caller ID, call waiting, voice mail,
    E911, billing, etc.
  • Ex Lucent 5ESS, Nortel DMS, Siemens EWSD
  • Class 4
  • Located in the Core of the network
  • Trunk to Trunk
  • Aprox. 800 deployed
  • Services call routing, screening, 800 services,
    calling cards, etc.
  • Ex Lucent 4ESS, Nortel DMS, Siemens

49
The PSTN NANP
  • NANP North American Numbering Plan
  • 3 digits area code 3 digits office code 4
    digits phone
  • Each Local Exchange Carrier (LEC) switch are
    assigned a block of at least 10,000 numbers
  • The Inter-Exchange Carrier (IXC) switches are
    responsible for transmitting long distance

PSTN
(212) 555 4210
50
The PSTN Call Routing
  • Both NANP and International Numbering Plan
    E.164, use prefix-based dialing

408
555
PSTN
51
Telephone System Summary
  • Analog narrowband circuits home-gt central office
  • 64 kb/s continuous transmission, with compression
    across oceans
  • ?-law 12-bit linear range -gt 8-bit bytes
  • Everything clocked a multiple of 125 s
  • Clock synchronization ? framing errors
  • ATT 136 tollswitches in U.S.
  • Interconnected by T1, T3 lines SONET rings
  • Call establishment out-of-band using
    packet-switched signaling system (SS7)

52
Telecommunications Regulation History
  • FCC regulations cover telephony, cable, broadcast
    TV, wireless etc
  • Common Carrier provider offers conduit for a
    fee and does not control the content
  • Customer controls content/destination of
    transmission assumes criminal/civil
    responsibility for content
  • Local monopolies formed by ATTs acquisition of
    independent telephone companies in early 20th
    century
  • Regulation forced because they were deemed
    natural monopolies (only one player possible in
    market due to enormous sunk cost)
  • FCC regulates interstate calls and state
    commissions regulate intra-state and local calls
  • Bells 1000 independents interconnected
    expanded

53
Deregulation of telephony
  • 1960s-70s gradual de-regulation of ATT due to
    technological advances
  • Terminal equipment could be owned by customers
    (CPE) gt explosion in PBXs, fax machines,
    handsets
  • Modified final judgement (MFJ) breakup of ATT
    into ILECs (incumbent local exchange carrier) and
    IXC (inter-exchange carrier) part
  • Long-distance opened to competition, only the
    local part regulated
  • Equal access for IXCs to the ILEC network
  • 1 long-distance number introduced then
  • 800-number portability switching IXCs gt retain
    800 number
  • 1995 removed price controls on ATT

54
US Telephone Network Structure (after 1984)
Eg ATT, Sprint, MCI
Eg SBC, Verizon, BellSouth
55
Telecom Act of 1996
  • Required ILECs to open their markets through
    unbundling of network elements (UNE-P),
    facilities ownership of CLECs.
  • Today UNE-P is one of the most profitable for
    ATT and other long-distance players in the local
    market due to apparently below-cost regulated
    prices
  • ILECs could compete in long-distance after
    demonstrating opening of markets
  • Only now some ILECs are aggressively entering
    long distance markets
  • CLECs failed due to a variety of reasons
  • But long-distance prices have dropped
    precipitously (ATTs customer unit revenue in
    2002 was 11.3 B compared to 1999 rev of 23B)
  • ILECs still retain over 90 of local market
  • Wireless substitution has caused ILECs to develop
    wireless business units
  • VoIP driven cable telephony wireless telephony
    gt more demand elasticity for local services

56
VoIP Technologies
57
IP Telephony Protocols SIP, RTP
  • Session Initiation Protocol - SIP
  • Contact office.com asking for bob
  • Locate Bobs current phone and ring
  • Bob picks up the ringing phone
  • Real time Transport Protocol - RTP
  • Send and receive audio packets

58
Inside the Endpoint Data-plane
  • I.e.after signaling is done
  • Consists of three components

User
User speaks into microphone, either PC attached,
regular analogue phone or IP phone
A/D Codec
Device digitizes voice according to certain
codecs G.711 / G.723.1 / G.729 ...
IP Gateway
Voice gets transmitted via RTP over an IP
infrastructure
59
Internet Multimedia Protocol Stack
60
Packet Encapsulation
RTP datagram
Version, flags CC
SequenceNumber
Timestamp
SynchronizationSource ID
PayloadType
CSRC ID (if any)
Codec Data
1 1 2
4
4 0-60
0-1460
UDP datagram
SourcePort Number
DestinationPort Number
UDP length
UDP checksum
Data
2
2
2 2
0-1472
Version header length
Protocol
IP packet
DestinationAddress
SourceAddress
HeaderChecksum
Data
Flags Frag Offset
TotalLength
PacketID
Options(if any)
TOS
TTL
1 1 2 2
2 1 1 2
4 4
0-40 0-1480
Start of frame delimiter
Length or Ethertype
Ethernet Frame
Preamble
DestinationAddress
Data
SourceAddress
Inter-frame gap
Pad
Checksum
12 7
1 6
6 2 0-1500
0-46 4
61
RTP Real-time Transport Protocol
RTP datagram
Version, flags CC
SequenceNumber
Timestamp
SynchronizationSource ID
PayloadType
CSRC ID (if any)
Codec Data
1 1 2
4
4 0-60
0-1460
  • Byte 1 Version number, padding yes/no, extension
    y/n, CSRC count
  • Byte 2 Marker, Payload type
  • Bytes 3,4 Sequence number for misordered and
    lost packet detection
  • Bytes 5-8 Timestamp of first data octet for
    jitter calculation
  • Bytes 9-12 Random syncronization source ID
  • Bytes 13-x Contributing Source ID for payload
  • Codec Data the actual Voice or Video bytes

62
RTCP Real-time Transport Control Protocol
  • RTCP is sent between RTP endpoints periodically
    to provide
  • Feedback on quality of the call by sending
    jitter, timestamps, and delay info back to sender
  • Carry a persistent transport-level identifier
    called the canonical name (CNAME) to keep track
    of participants and synchronize audio with video
  • Carry minimal session information (like
    participant IDs), although signaling protocols do
    this much better
  • RTCP is mandatory for multicast sessions and for
    many point-to-point protocols, but some boxes
    dont implement it
  • Uses another UDP port (usually RTPs port 1)

63
SIP
64
Signaling VoIP Camps
Circuit switch engineers We over IP
Convergence ITU standards
Conferencing Industry
Netheads IP over Everything
H.323
SIP
Softswitch
BICC
ISDN LAN conferencing
I-multimedia WWW
Call Agent SIP H.323
BISDN, AIN H.xxx, SIP
IP
IP
IP
any packet
65
H.323 vs SIP
  • H.323 ITU standard
  • Derived from telephony protocol (Q.931)
  • Follows ISDN model same control message
    sequences
  • Interfaces well with telephony services (H.450,
    Q.SIG)
  • SIP IETF standard
  • Derived from HTTP style signaling,
  • Simple and interfaces well with IP networks,
    instant messaging (IM)
  • Services are not explicitly exposed to protocol
  • Well-defined methods can be used to design
    services most telephony services have analogs in
    the SIP world today
  • SIP is gathering market share rapidly

66
SIP
Audio Codec G.711 G.723 G.729
Video Codec H.261 H.263
RTP
RTCP
SIP
TCP
UDP
IP
LAN Interface
67
SIP functionality
  • IETF-standardized peer-to-peer signaling protocol
    (RFC 2543)
  • Locate user given email-style address
  • Setup session (call)
  • (Re)-negotiate call parameters
  • Manual and automatic forwarding
  • Personal mobility different terminal, same
    identifier
  • Call center reach first (load distribution) or
    reach all (department conference)
  • Terminate and transfer calls

68
SIP Addresses Food Chain
69
Why is SIP interesting?
  • SIP is IETFs equivalent for H.323 to provide a
    peer-based signaling protocol for session setup,
    management and teardown
  • Simple, did not inherit the complexity of ISDN
  • Analogy CISC architecture
  • Though all services arent defined as in H.323,
    you can compose them with primitives
  • Was designed with multimedia in mind
  • Just requires a MIME type
  • Tremendous flexibility can add video, text etc
    to a voice session, similar to what HTTP did to
    Internet content
  • Like H.323, can use SIP end-to-end with no
    network infrastructure (MGC etc.) peer-to-peer
  • Lightweight ? can be embedded in small devices
    like handhelds

70
IP SIP Phones and Adaptors
1
  • Are true Internet hosts
  • Choice of application
  • Choice of server
  • IP appliances
  • Implementations
  • 3Com (3)
  • Columbia University
  • MIC WorldCom (1)
  • Mediatrix (1)
  • Nortel (4)
  • Siemens (5)

Analog phone adaptor
2
3
Palm control
4
5
4
71
SIP Personal Mobility
Users maintain a single externally visible
identifier regardless of their network location
72
Expand existing PBXs w/ IP phones
  • Transparently

73
SIP as Event Notification Protocol
74
SIP Presence
75
Light-weight signaling Session
InitiationProtocol (SIP)
  • IETF MMUSIC working group
  • Light-weight generic signaling protocol
  • Part of IETF conference control architecture
  • SAP for Internet TV Guide announcements
  • RTSP for media-on-demand
  • SDP for describing media
  • others malloc, multicast, conference bus, . . .
  • Post-dial delay 1.5 round-trip time (with UDP)
  • Network-protocol independent UDP or TCP (or AAL5
    or X.25)

76
SIP components
  • UAC user-agent client (caller application)
  • UAS user-agent server accept, redirect, refuse
    call
  • redirect server redirect requests
  • proxy server server client
  • registrar track user locations
  • user agent UAC UAS
  • often combine registrar (proxy or redirect
    server)

77
SIP-based Architecture
78
Example Call
  • Bob signs up for the service from the web as
    bob_at_ecse.rpi.edu
  • sipd canonicalizes the destination to
    sipbob_at_ecse.rpi.edu
  • He registers from multiple phones
  • sipd rings both ephone and sipc
  • Bob accepts the call from sipc and starts talking
  • Alice tries to reach Bob
  • INVITE ipBob.Wilson_at_ecse.rpi.edu

ecse.rpi.edu
79
SIP Sessions
  • Session exchange of data between an
    association of participants
  • Users may move between endpoints
  • Users may be addressable by multiple names
  • Users may communicate in several different media
  • SIP enables internet endpoints to
  • Discover each other
  • Characterize the session
  • Location infrastructure proxy servers,
    invite/register
  • Name mapping and redirection services
  • Add/remove participants from session
  • Add/remove media from session

80
SIP Capabilities
  • User location determination of the end system to
    be used for communication
  • User availability determination of the
    willingness of the called party to engage in
    communications
  • User capabilities determination of the media and
    media parameters to be used
  • Session setup "ringing", establishment of
    session parameters at both called and calling
    party
  • Session management including transfer and
    termination of sessions, modifying session
    parameters, and invoking services.

81
What SIP is not
  • SIP is not a vertically integrated communications
    system.
  • It is a component in a multimedia architecture.
  • SIP does not provide services.
  • Rather, SIP provides primitives that can be used
    to implement different services.
  • For example, SIP can locate a user and deliver an
    opaque object to his current location.
  • SIP does not offer conference control services
  • such as floor control or voting
  • SIP does not prescribe how a conference is to be
    managed.

82
SIP Structure
  • 3 layers, loosely coupled, fairly independent
    processing stages
  • Lowest layer syntax, encoding (augmented BNF)
  • Second layer transport layer.
  • Defines how a client sends requests and receives
    responses and how a server receives requests and
    sends responses over the network.
  • Third layer transaction layer.
  • A transaction is a request sent by a client
    transaction (using the transport layer) to a
    server transaction
  • along with all responses to that request sent
    from the server transaction back to the client.
  • The transaction layer handles application-layer
    retransmissions, matching of responses to
    requests, and application-layer timeouts
  • The layer above the transaction layer is called
    the transaction user (TU).

83
SIP Design Choices
84
Proxy Server
Location Server
Proxy server
85
Redirect Server
us.gov
Location Server
parliament.uk
tony_at_parliament.uk
Redirect Server
86
SIP Call Signaling
Assumes Endpoints(Clients) know each others IP
addresses
SIP Endpoint
SIP Gateway
Invite
Signaling Plane
SIP SDP (TCP or UDP)
180 Ringing
200 OK
Ack
RTP Stream
Bearer Plane
Media (UDP)
RTP Stream
RTCP Stream
87
PSTN to IP Call
88
IP to PSTN Call
89
Traditional voice mail system
Bob can listen to his voice mails by dialing some
number.
90
SIP-based Voicemail Architecture
vm.office.com
The voice mail server registers with the SIP
proxy, sipd
Alice calls bob_at_office.com through SIP proxy.
SIP proxy forks the request to Bobs phone as
well as to a voicemail server.
91
Voicemail Architecture
v-mail
vm.office.com
After 10 seconds vm contacts the RTSP server for
recording.
vm accepts the call.
Sipd cancels the other branch and ...
rtspd
...accepts the call from Alice.
Now user message gets recorded
92
IETF SIP Architecture Tour Roundup
Registrar Proxy or Redirect Server
Gateway
PSTN, ISDN, ATM, etc
User Agent
User Agent
User Agent
Endpoints
Media streams RTP/RTCP (G.911, G.723.1, )
93
IETF SIP Architecture Tour Roundup
Registrar Proxy or Redirect Server
Gateway
PSTN, ISDN, ATM, etc
  • System Management
  • admission control
  • address translation/forwarding
  • Firewall bypassing

User Agent
User Agent
User Agent
Interface to non-IP or H.323 networks
Endpoints
Media streams RTP/RTCP (G.911, G.723.1, )
Conferencing does not need another box (MCU)
End-user devices and network proxies
94
IETF SIP Architecture Tour Roundup
Registrar Proxy or Redirect Server
Gateway
PSTN, ISDN, ATM, etc
User Agent
User Agent
User Agent
Endpoints
Media streams RTP/RTCP (G.911, G.723.1, )
  • Components of the SIP protocol suite
  • SIP almost all signaling, optional services,
    etc.
  • SDP negotiation/capabilities
  • DNS address translation
  • RSVP QoS bandwidth guarantee

95
SDP Session Description Protocol
  • Not really a protocol describes data carried by
    other protocols
  • Used by SAP, SIP, RTSP, H.332, PINT. Eg
  • v0
  • og.bell 877283459 877283519 IN IP4 132.151.1.19
  • sCome here, Watson!
  • uhttp//www.ietf.org
  • eg.bell_at_bell-telephone.com
  • cIN IP4 132.151.1.19
  • bCT64
  • t3086272736 0
  • kclearmanhole cover
  • maudio 3456 RTP/AVP 96
  • artpmap96 VDVI/8000/1
  • mvideo 3458 RTP/AVP 31
  • mapplication 32416 udp wb

96
Upcoming SIP Extensions (probable)
  • Call Admission Control
  • Caller Preferences and Callee Capabilities
  • Call Transfer
  • SIP to ISUP mapping
  • SIP to H.323 mapping
  • Resource Management (QoS preconditions)
  • Caller/Callee Name Privacy
  • SIP Security
  • Supported Options Header
  • Session Timer Refresh
  • Distributed Call State
  • 3rd Party Call Control
  • Early media for PSTN interoperability
  • There are currently 47 drafts in the pipeline!
  • 174 Drafts have expired

97
SIP Dialogs (RFC 3261)
  • A dialog represents a peer-to-peer SIP
    relationship between two user agents that
    persists for some time.
  • The dialog facilitates sequencing of messages
    between the user agents and proper routing of
    requests between both of them.
  • The dialog represents a context in which to
    interpret SIP messages.
  • A dialog is identified at each UA with a dialog
    ID, which consists of a Call-ID value, a local
    tag and a remote tag.
  • A dialog contains certain pieces of state needed
    for further message transmissions within the
    dialog.
  • Note dialog is within SIP whereas sessions are
    outside SIP

98
UPDATE method (RFC 3311)
  • INVITE method initiation and modification of
    sessions.
  • INVITE affects two pieces of state session (the
    media streams SIP sets up) and dialog (the state
    that SIP itself defines).
  • Issue need to modify session aspects before the
    initial INVITE has been answered.
  • A re-INVITE cannot be used for this purpose
    impacts the state of the dialog, in addition to
    the session.
  • Ans The UPDATE method
  • Operation (Offer/Answer model)
  • The caller begins with an INVITE transaction,
    which proceeds normally.
  • Once a dialog is established, either early or
    confirmed,
  • the caller can generate an UPDATE method that
    contains an SDP offer for the purposes of
    updating the session.
  • The response to the UPDATE method contains the
    answer.
  • Similarly, once a dialog is established, the
    callee can send an UPDATE offer

99
Locating SIP Servers (RFC 3263)
  • UA ? Proxy ? Remote Proxy ? UA
  • I.e Go via proxies (per-domain)
  • Issue need to locate remote proxy (use DNS)
  • DNS NAPTR (type of server) and SRV (server URL)
    queries are used to locate the specific servers.
  • Different transport protocols can be used
    (TLSTCP, TCP, UDP, SCTP)

100
SIP for instant messaging IM (RFC 3428)
  • IM transfer of (short) messages in near
    real-time, for conversational mode.
  • Current IM proprietary, server-based and linked
    to buddy lists etc
  • MESSAGE method inherits SIPs request routing
    and security features
  • Message content as MIME body parts
  • Sent in the context of some SIP dialog
  • (note slightly different from pager mode
    asynchronous)
  • Sent over TCP (or congestion controlled
    transports) lots of messaging volumes
  • Allows IM applications to potentially
    interoperate and also provides SIP-based
    integration with other multimedia streams.

101
SIP compression (RFC 3486)
  • Cannot use DNS SRV and NAPTR techniques
    non-scalable (only useful for specifying
    transport protocol options)
  • Use an application-level exchange to specify
    compression of signaling info
  • sipalice_at_atlanta.comcompsigcomp
  • Via SIP/2.0/UDP server1.foo.com5060branchz9hG4
    bK87a7compsigcomp
  • SIGCOMP is the compression protocol

102
Device Configuration
103
SIP Scaling Issues
104
SIP Scaling (contd)
SIP Load Characteristics
105
H.323
106
SIP vs H.323 vs Megaco
107
H.323 vs SIP
Typical UserAgent Protocol stack for Internet
Terminal Control/Devices
Terminal Control/Devices
Q.931
H.245
RTCP
RAS
RTCP
SIP
SDP
Codecs
Codecs
RTP
RTP
TPKT
TCP
UDP
Transport Layer
IP and lower layers
108
SIP versus H.323
H.323 and SIP are direct competitors in
peer-level call control space
109
SIP-H.323 Interworking ProblemsEg Call setup
translation
H.323
SIP
Q.931 SETUP
INVITE
Destination address (Bob_at_office.com)
Q.931 CONNECT
200 OK
Terminal Capabilities
Media capabilities (audio/video)
Terminal Capabilities
ACK
Open Logical Channel
Media transport address (RTP/RTCP receive)
Open Logical Channel
  • H.323 Multi-stage dialing

110
H.323 Standard Series
Audio Codec G.711 G.723 G.729
Video Codec H.261 H.263
System Control
H.245 Control
H.225 Call Setup
RTP
RTCP
RAS Gatekeeper
TCP
UDP
IP
LAN Interface
111
Internet Telephony Protocols H.323
112
H.323 (contd)
  • Terminals, Gateways, Gatekeepers, and Multipoint
    Control Units (MCUs)

113
H.323 Model - Gatekeeper Routed Call
Gatekeeper
RAS
Call Setup/Signaling
RAS
Call Setup/Signaling
Call Control
Call Control
Voice Channel
Endpoint
Gateway
114
H.323 Model - Gatekeeper Direct Call
Gatekeeper
RAS
RAS
Call Setup/Signaling
Call Control
Voice Channel
Endpoint
Gateway
115
H.323 Call Signaling
Assumes Endpoints(Clients) know each others IP
addresses
H.323 Endpoint
H.323 Gateway
Setup
H.225 (TCP) (Q.931)
Alerting
Connect
Terminal Capability Set
Signaling Plane
Terminal Capability Set Acknowledge
Terminal Capability Set Acknowledge
H.245 (TCP)
Open Logical Channel
Open Logical Channel Acknowledge
Open Logical Channel Acknowledge
RTP Stream
Bearer Plane
Media (UDP)
RTP Stream
RTCP Stream
H.323v1 (5/96) - 7 or 8 Round Trips H.323v2 Fast
Start (2/98) - 2 Round Trips
116
ITU-T H.323 Architecture Tour
Gate Keeper (GK)
Gateway (GW)
PSTN, ISDN, ATM, etc
Terminal
Terminal
Terminal
Endpoints
Media streams RTP/RTCP (G.911, G.723.1, )
117
ITU-T H.323 Architecture Tour
Gate Keeper (GK)
Gateway (GW)
PSTN, ISDN, ATM, etc
  • System Management
  • zone management
  • b/w management admission control
  • address translation
  • centralized control (gatekeeper control mode)

Terminal
Terminal
Terminal
Interface to non-IP networks
Endpoints
Media streams RTP/RTCP (G.911, G.723.1, )
Conferencing
End-user devices and network proxies
118
ITU-T H.323 Architecture Tour
Gate Keeper (GK)
Gateway (GW)
PSTN, ISDN, ATM, etc
Terminal
Terminal
Terminal
Endpoints
Media streams RTP/RTCP (G.911, G.723.1, )
119
Gatekeeper Routed Call
1. Setup called 5551234 caller
964274910.0.0.5
2. Setup called 5551234192.168.0.3 caller
9642749
3. Connect
Atlanta Zone (404)
2, 6, 10, 14
1, 5, 9, 13
Gatekeeper 132.177.120.5
223-2749 10.0.0.5
3, 7, 11, 15
4, 8, 12, 16
4. Connect
9. Open Channel G.729/30ms, 10.0.0.56400
5. TCS media G.711/30ms, G.729/30ms
10. Open Channel G.729/30ms, 10.0.0.56400
6. TCS media G.711/30ms, G.729/30ms
11. Open Channel G.729/20ms, 192.168.0.32300
7. TCS media G.729/20ms, G.723
12. Open Channel G.729/20ms, 192.168.0.32300
8. TCS media G.729/20ms, G.723
13. ACK
14. ACK
15. ACK
16. ACK
120
Gatekeeper Direct Call
1. ARQ called 5551234 caller
964274910.0.0.5
2. ACF called 5551234192.168.0.3
3. Setup called 5551234 caller
964274910.0.0.5
Atlanta Zone (404)
1
Gatekeeper 132.177.120.5
2
223-2749 10.0.0.5
3, 5, 7, 9
4, 6, 8, 10
4. Connect
9. ACK
5. TCS media G.711/30ms, G.729/30ms
10. ACK
6. TCS media G.729/20ms, G.723
7. Open Channel G.729/30ms, 10.0.0.56400
8. Open Channel G.729/20ms, 192.168.0.32300
121
MEGACO/H.248, Softswitch Concepts
122
Master/Slave vs. Peer Comparison
Master/Slave (Thin Client)
Peer (Thick Client)
Protocols
  • MEGACO/H.248, MGCP
  • H.323, SIP

123
Megaco/H.248
Audio Codec G.711 G.723 G.729
Video Codec H.261 H.263
RTP
RTCP
Megaco
TCP
UDP
IP
LAN Interface
124
Megaco/H.248 Convoluted History
PacketCable Network-based Call Signaling (NCS)
based on earlier version of MGCP (March 99)
DSM-CC
Diameter
Industry Defacto Std.
PacketCable NCS
IPDC
MGCP (proposal)
I-RFC 2705
Non-Standard
SGCP
MGCP proposal
MGCP released as Informational RFC (Oct 99)
MDCP (proposal)
Not fully accepted by Megaco WG, diverged (Spring
99)
Megaco Protocol
Megaco Protocol stream created, true consensus
(March 99)
ITU H.GCP
WORLD STANDARD
Megaco/H.248
ITU SG-16 initiates gateway control project,
H.GCP starting from MDCP (May 99)
Agreement reached between ITU SG16 and IETF
Megaco to work together to create one standard
(Summer 99)
125
Megaco Vs MGCP
Megaco/H.248
Call Model Termination Context Topology
P2P Single Media Single Media Conferencing
P2P Multimedia Multimedia
Conferencing Terminations Physical
Ephemeral Muxing Template
Command Grouping Transaction Events
Event Buffering Event Packages (MGCP
Packages Additional
Packages) National Variants Media
Session Description SDP H.245
Protocol Encoding Binary
Text Transport TCP UDP SCTP Security
Authentication Header MGC Backup
MGCP
Event Packages (MGCP) Media Session
Description SDP Protocol Encoding
Text Transport UDP
Call Model Termination Connection
P2P Single Media Single Media
Conferencing Terminations Physical
Ephemeral Command Grouping Ad hoc
Embedding Event Quarantine
Bold entries indicate additional features in
Megaco vs. MGCP
126
Megaco Architecture Whirlwind Tour
  • Signalling Gateway Layer (SG)
  • Interface to SS7 signalling etc
  • Not in Megaco scope (IETF Sigtran)
  • Media Gateway Control Layer (MGC)
  • Contains all call control intelligence
  • Implements call level features (forward,
    transfer, conference, hold, )

Megaco Protocol
  • Media Gateway Control Protocol
  • Master / slave control of MGs by MGCs
  • Connection control
  • Device control and configuration
  • Orthogonal to call control protocols
  • Media Gateway Layer (MG)
  • Implements connections to/from IP cloud (through
    RTP)
  • Implements or controls end device features
    (including UI)
  • No knowledge of call level features

127
Framework for H248/Megaco Protocol
  • Media Gateway
  • Connection and device control
  • No call processing, no call model
  • Service-independent
  • Cost effective
  • Media GW Controller
  • Call processing and Service logic
  • Call routing
  • Inter-peer entity communication via call control
    protocols (e.g. H.323, SIP, etc)

Device control
Media Gateway
Device control
PBX/CO
PBX/ CO
PSTN trunking Media Gateway
PBX Media Gateway
PSTN line Media Gateway
Telephone/Residential Media Gateway
IP (or ATM) Network
IP Phone Media Gateway
128
Megaco Framework
  • The MGC and MGs form a virtual IP-based switch
  • Looks like an H.323 Gateway to other H.323
    devices, and a SIP Server to other SIP devices
  • RTP (the voice media itself) is still
    point-to-point

Media Gateways
PSTN Trunking Media Gateway
Media GW Controller
PSTN Line Media Gateway
Megaco/ H.248
Telephone/Residential Media Gateway
Cable Modem Media Gateway
129
Megaco call in action (optional)
MG2
MG1
MGC
Powered On
Powered On
ServiceChange Restart
ServiceChange Restart
Reply ServiceChange
Reply ServiceChange
Modify Look for Off-Hook
Modify Look for Off-Hook
Ready
Ready
Reply Modify
Reply Modify
Off-Hook
Notify Off-Hook
Reply Notify
Dial Tone, User Dials
Modify Dial Tone, Digit Map
Reply Modify
Notify number 19782886160
Reply Notify
Add TDM to RTP, what codecs?
Reply Add, codec G.729
130
Megaco call in action (continued)
MG2
MG1
MGC
Add TDM to RTP, ring phone
Phone Rings
Reply Add
Modify ip of MG2, ringback
Hears Ring
Off-Hook
Reply Modify
Notify Off-hook
Reply Notify
Modify stop ring
Stops Ring
Reply Modify
Modify stop ringback, fullduplex
Reply Modify
Open RTP
Open RTP
Active Call/End of Invite Request
On-Hook
Notify On-hook
Reply Notify
SubtractTDM and RTP
Subtract TDM and RTP
Disconnect
Reply Subtract
Reply Subtract
131
Megaco/H.248 IP Phone Control
Ciscos Skinny, Nortels UNIStim, etc., are very
similar protocols but theyre not interoperable
H.323 GW
MGC
H.323
Voice (RTP)
In theory the RTP stream should go direct
phonelt-gtGW, but many today tandem through the MGC
Media, LCD, Softkey Control
Media, LCD, Softkey Control
Voice (RTP)
Voice (RTP)
Voice (RTP)
Voice (RTP)
IP Phone Media Gateway
IP Phone Media Gateway
132
Vendor Support for Standards
  • Source Network World and Mier Communications -
    August, 2001

133
H.323 limitations
  • Gateway did a lot of things that were easily
    decomposed into functionally complete pieces
  • Key insight from layering separate functionally
    complete pieces as far as possible.
  • Quickly faced scaling problems
  • Call setup and control was a complex control
    plane operation
  • Media translation between a variety of networks
  • Take-away point ? Build a distributed system that
    acts as a single logical entity to the user

134
MGCP/H.248/Megaco
SIP
Media Gateway Controller (MGC)
Media Gateway Controller (MGC)
Master/Slave
MGCP
Media Gateway
Media Gateway
Signaling Gateway
Signaling Gateway
Distributed entities acting in co-ordination
Connect to variety of networks, home users and
other media receptors like H.323 terminals etc
Interface to variety of signaling mechanisms
Separate signaling and voice planes, but user
unaware of it
User A
For examples of gateways see RFC 3435
135
Softswitch Motivation
Class-4/5 switches bulky, expensive. Incentive to
switch to cheaper easily managed IP
PSTN
Class 4 switch
Class 5 switch
Voice
Class 5 switch
Users
Users
ISDN Switch
H.323 gateway
Data
Initial gateway between PSTN and Internet was
H.323. Gateway did signaling, call control,
translation in one box. Not scalable.
Packet networks
136
What is a Softswitch?
  • A Softswitch is a device independent software
    platform designed to facilitate telecommunication
    services in an IP network
  • A Softswitch controls the network
  • At a high level, a Softswitch is responsible for
  • Protocol Conversion
  • Control and synchronization of Media Gateways
  • Its an Architecture, NOT a box

137
The softswitch concept
  • Build a distributed system that performs the
    functions of the Class-4/5 switches
  • Use generic computing platforms to reduce cost,
    size and flexibility
  • E.g., DSPs or other programmable architectures
  • Software components to implement many of the
    switching tasks give the soft part of
    softswitch
  • The MGC which does the call control and is the
    brain of the system is usually referred to as the
    softswitch or call agent
  • The gateways are dumb devices which do whatever
    MGC instructs them to do
  • MGC therefore does
  • Call setup, state maintenance, tear-down
  • Megaco was an earlier non-standard framework
    which was later standardized jointly by ITU and
    IETF as MGCP

138
Softswitch Whats the big deal?
  • Unprecedented flexibility
  • Smaller offices can have just gateways, MGCs can
    be at some remote data center
  • Standards-based interactions drive down costs and
    offer wider architectural choices
  • Fast introduction of services and applications
    that can again be located remotely only need
    MGCs to upgrade
  • New hosted-services solutions due to flexibility
  • Dramatic space savings
  • Sometimes as much as 10 times smaller even with
    all the components of the softswitch architecture

139
Softswitch Architecture
Application Server
  • Distributed functionality
  • Open platforms
  • Open interfaces enable new services
  • Leverages the intelligence of endpoints
  • Media agnostic

Media Gateway Controller
Signaling Gateway
Media Gateway
PSTN/ End users
140
Softswitch - Media Gateway Controller
An SS7 Enabled Media Gateway Controller
integrates the functionality of new applications
with the large installed based of legacy systems.
Application Server
  • Multiple controllers can collaborate on a single
    call
  • May be distributed across the globe
  • May or may not be collocated with SS7 Signaling
    Gateway

Media Gateway Controller
Signaling Gateway
Media Gateway
PSTN/ End users
141
Softswitch - Media Gateway Controller Functions
Application Server
  • Connections (call setup and teardown)
  • Events (detection and processing)
  • Device management (gateway startup, shutdown,
    alerts)

Media Gateway Controller
Signaling Gateway
Media Gateway
PSTN/ End users
142
Softswitch - Media Gateways
Media Gateways provide interaction between audio
in the network and software controlled
applications
Application Server
  • Convert PSTN to IP packets
  • Convert IP packets to PSTN
  • In-band event detection and generation
  • Compression (G.7xx,)
  • May be distributed across the globe

Media Gateway Controller
Signaling Gateway
Media Gateway
PSTN/ End users
143
MGC and MG Roles
  • Media Gateway Controller
  • MGCs allow intelligence to be distributed in the
    network
  • Basic call routing functions
  • Synchronization of Media Gateways
  • Protocol Conversion
  • Media Gateway
  • MGs are purpose built specialist devices
  • Trunking gateways
  • VoATM gateways
  • Access gateways
  • Circuit switches
  • Network Access Servers

144
Softswitch - Signaling Gateway
Signaling Gateways provide interaction between
the SS7 network and Media Gateway Controllers.
Application Server
  • Convert SS7 to IP packets
  • Convert IP to SS7 packets
  • Signaling transport (SS7, SIP-T, Q.931)
  • Extremely secure
  • Extremely fault tolerant

Media Gateway Controller
Signaling Gateway
Media Gateway
PSTN/ End users
145
Softswitch Application Server
Application Servers(AS) provide the new services
that are the real value-add for Softswitches.
Application Server
  • Many core features are part of the MGC
  • Allows new features to be developed by third
    parties

Media Gateway Controller
Signaling Gateway
Media Gateway
PSTN/ End users
146
Softswitch Application Server
Application Servers(AS) Can be broken apart and
distributed in the network
LDAP
Feature Server
Directory Server
COPS
Corba
Network Elements
Policy Server
SIP
Corba
Media Server
Management Server
Connectivity Server
SIP,Parlay,JAIN
147
Softswitch Architecture The protocols
Application Server
SIP, Parlay, Jain
Media Gateway Controller
Sigtran w/SCTP
Signaling Gateway
H.248,MGCP
Media Gateway
PSTN/ End users
148
Softswitch Architecture Interdomain protocols
Application specific
Application Server
Application Server
SIP, Parlay, Jain
Media Gateway Controller
Media Gateway Controller
Sigtran
SIP-T,BICC
Signaling Gateway
Signaling Gateway
H.248,MGCP
Media Gateway
Media Gateway
RTP
PSTN/ End users
PSTN/ End users
149
SIP vs MEGACO Summary
150
SIP vs MEGACO (contd)
151
VoIP Signaling Model Summary
  • End-system SIP signaling (beat out H.323)
  • PSTN gateway, with interfaces looking into PSTN
    and interfaces looking into VoIP networks
  • Media Gateway Controller (MGC) intelligent
    endpoint supervises call services end-end
  • Media Gateway (MG) interface to the IP network
    or PSTN simple endpoint instructed by MGC
  • MEGACO MG ? MGC interaction protocol
  • ITU (H.248) and IETF (RFC 3525) standard
  • Replaces proprietary APIs and RFC 3435 (MGCP)

152
Speech Coding and Speech Coders for VoIP
153
Taxonomy of Speech Coders
  • Waveform coders attempts to preserve the
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