internet telephony: voip, sip & more
DESCRIPTION
Internet Telephony: VoIP, SIP & more. Shivkumar Kalyanaraman. : “ shiv rpi ”. Adapted from slides of Henning Schulzrinne, Doug Moeller. Overview. Telephony: history and evolution IP Telephony: What, Why & Where? Adding interactive multimedia to the web - PowerPoint PPT PresentationTRANSCRIPT
Shivkumar KalyanaramanRensselaer Polytechnic Institute
1
Internet Telephony: VoIP, SIP & more
: “shiv rpi”
Shivkumar Kalyanaraman
Adapted from slides of Henning Schulzrinne, Doug Moeller
Shivkumar KalyanaramanRensselaer Polytechnic Institute
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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
Overview
Shivkumar KalyanaramanRensselaer Polytechnic Institute
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What is VoIP? Why VoIP?
Where is VoIP Today?
Shivkumar KalyanaramanRensselaer Polytechnic Institute
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What is VoIP?
VoIP = “Voice over IP” Transmission of telephony services via IP infrastructure => 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
Shivkumar KalyanaramanRensselaer Polytechnic Institute
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What is VoIP? Protocol Soup
H.323
SIPMGCPSGCP
H.GCP
Megaco
IPD
C
H.245
SDP
MDCP
SigtransVPIM
Q.SIG
“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 year’s model.” [Tanenbaum]
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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
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Why VoIP? Telephony: Mature IndustryAT&T Divestiture
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Why VoIP: Price/call plummeting due to overcapacity
AT&T Divestiture
1996 deregulation
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Relevant Telecom Industry Trends 1984: AT&T 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 => cell phone instead of land-lines
Developing countries leap frog to cell phones 3G, WiMax => broadband, VoIP & mobility
Broadband rollouts happening slowly, but picking up steam now. Cable offering converged & bundled services:
digital cable, VoIP, video Recent mergers: AT&T (long-distance & data network provider) bought by
SBC (baby bell); Verizon/Qwest vs MCI saga…
Shivkumar KalyanaramanRensselaer Polytechnic Institute
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Why VoIP ? Data vs Voice Traffic
Infrastructure convergence: Since we are building future networks for data, can we slowly junk the voice infrastructure and move over to IP?
Note: quantity quality value-added
Interactive svcs (phone, cell, sms)still dominate on a $$-per-Mbps basis
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Trends: Total Phone vs Data Revenues
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Motivations and drivers
Class 5switch
Class 4switch
Class 5switch
UsersUsers
PSTN
Packet networks
Data
Voice
H.323 gateway
ISDN Switch
Class-4/5 switches bulky, expensive. Incentive to switch to cheaper easily managed IP
Initial gateway between PSTN and Internet was H.323. Gateway did signaling, call control, translation in one box. Not scalable.
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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
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VoIP Volumes Are Accelerating While Adoption of Applications is Growing
020000400006000080000
100000120000140000160000180000200000
2001 2002 2003 2004 2005 2006 2007
VoIP VPN Traffic
Source: Probe Research Inc.: Reaching the Big Guys + Global Enterprise Forecast. September 2002
0
50000
100000
150000
200000
250000
300000
350000
2001 2002 2003 2004 2005 2006 2007
Virtual PBX + Managed IP PBX traffic
M ofMinutes
M ofMinutes
North America Rest of the World
North America Rest of the World
Enterprise Adoption of VoIP / IPT Applications
• 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: Giga Group, "Next Generation IP Telephony Applications Deliver Strategic Business Value", October 20, 2003
Respondents
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Drivers Are Evolving From Cost Savings to Added Business Value…
Gartner Group, Sept. 16, 2003
Business Case
Justification Based on Business
Value
V3 Apps
V3 Apps - e.g. Unified Communications, Application Integration With Communications
V2 Apps
Business Case Justification
Based on Investment Protection
V2 Apps - e.g. Call Center Functions, Messaging, Administration Tools and Reports
Percentage IP Phones Performing Functions Other Than
POTS
2003 2007200620052004 20082002
Business Case
Justification Based on
Cost Savings
V1 Apps
V1 Apps - e.g. IP-PBX, Basic Call Functions, Branch offices, Toll-bypass
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• End-to-End Interoperability
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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”
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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: AT&T 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
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Consumer VoIP over broadband
Broadband Infrastructure
ResidentialMedia Gateway
Traditional phone
Media Gateway Controller
Signaling and media gatewaysTo reach PSTN or other networks
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Consumer VoIP at home with cable
PacketCable standard with DOCSIS 1.1 access infrastructure
MTA(Media Terminal Adapter)
Cable Modem
Call Management Server
MGCCable Modem Term. Sys.
MediaGateway
SignalingGateway
Other access mechanisms will similarly hand over to an MGC
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Consumer VoIP: AT&T 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
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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
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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
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Enterprise: Private Branch Exchange (PBX)
7043
7040
7041
7042
External line
Telephoneswitch
Private BranchExchange
212-8538080
Anotherswitch
Corporate/Campus
InternetCorporate/Campus LAN
Post-divestiture phenomenon...
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Enterprise VoIP: Yesterday’s networksCircuit Switched Networks (Voice)
Packet Switched Networks (IP)
PBXPBX
COCO
CO
Router
RouterRouter
Router
Router
Headquarters Branch Offices
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Enterprise VoIP: Today’s networksToll by-pass
Circuit Switched Networks (Voice)
Packet Switched Networks (IP)
PBXPBX
COCO
CO
Router
RouterRouterRouter
Router
Headquarters Branch Offices
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Enterprise VoIP: Tomorrow’s networksUnified/Converged Networks
Unified Networks (Voice over IP)
Router
RouterRouter
Router
Router
COCO
Legacy PSTN
Headquarters Branch Offices
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• 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
• 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
AT&T’s Integrated Infrastructure Supports Multiple Endpoints, Access Technologies and Application Services
Common VoIP Connectivity Layer
H.323 endpoints PSTN
SIP endpoints
NG Border Elements
SIP Border Elements
H.323 Border
Elements
MGCP Border
Elements
MGCP endpoints
IP/MPLS Converged Network
AT&T Call Control Element
Voice Applications: IP Centrex, IP Call Center and Distant Worker
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Outbound Call• IP to Circuit Switched
NetworkAdjunct
Softswitch
App.Server Gateway
CustomerRecords
MediaServer
App.Server
Inbound Call• Circuit Switched to IP
800 Call• Circuit Switched to IP
Redirect Call• Circuit Switched to IP
SDN Call• IP to Circuit Switched
VoIP Network UtilitiesEnsure Seamless Operations
Circuit Switched Network
IP Network
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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
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Telephony-enabled packet networks
RouterVoIP
Gateway
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
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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
PBX
Switch
SwitchSwitch
PBXVFRAD
VFRAD
Router
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Voice over Packet: Market Forecast – North America
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Telephony: History, Review & Trends
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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]
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Public Telephony (PSTN) History 1876 invention of telephone 1915 first transcontinental telephone (NY–SF) 1920’s 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)
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PSTN Evolution
Full Mesh Office Switched
W/ HierarchyOffice Switched
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AT&T Telephony Hierarchy
1
109 8
7
3
2
45
6
1 2 3
1 2 3
1 2 3
65 66 67
228 229 230
1298 1299 1300
1 2 3 4 519,000
200 million telephones
19,000 endoffices
1300 tolloffices
230 primaryoffices
67 sectionaloffices
10 regionaloffices(full mesh)
Source: Computer Networks, Andrew S. Tanenbaum
Class 5
Class 4
Class 3
Class 2
Class 1
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PSTN early days 40s-60s
User AUser B
Local Office
Tandem Office
Local Office
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
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Advanced Intelligent Network
User AUser B
Voice Network
Local Office
Signaling Network
Customer Info forAdvanced services
•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
Sometimes also called Intelligent Network, arrival of services other than voice
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The PSTN – Architecture PSTN – Public Switched Telephone Network Uses digital trunks between Central Office switches (CO) Uses analog line from phones to CO
Digital Trunks
Analog line
CentralOffice (CO)
Analog Digital Analog
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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
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Quantization
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Companding
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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
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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
Channel 3
Channel 24
…
Channel1
Channel2
Channel3
Channel24…
Framing Bit
1 D4 Frame
TDM
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The PSTN – Architecture, Switches PSTN – Public Switched Telephone Network As the name says, it’s 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
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Legacy Digital Circuit Switch
• Centralized Intelligence
• Proprietary Code
• Proprietary service deployment
• Very expensive
TrunkCard
TrunkCard
TrunkCard
LineCard
LineCard
LineCard
Switch Controller
Next Switch
Next Switch
Next Switch
SS7 Network
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What’s 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
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PSTN
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
IXC212
LEC555
4210
(212) 555 4210
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The PSTN – Call Routing Both NANP and International Numbering Plan – E.164, use
prefix-based dialing
408 5644555212PSTN
1+212+555+5644
The first LEC receives a call, seeing ‘1’ as the first digit and then passing the call on to theIXC switch. The IXC then routes the call to the remote IXC responsible for ‘212’
555+5644
The ‘212’ IXC looks at the office code and passes it on to the ‘555’ LEC switch
5644
The ‘555’ LEC switch then checks the station code and signals the appropriate phone
SS7
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Telephone System Summary
Analog narrowband circuits: home-> central office 64 kb/s continuous transmission, with compression
across oceans-law: 12-bit linear range -> 8-bit bytes
Everything clocked a multiple of 125 s Clock synchronization framing errors
AT&T: 136 “toll”switches in U.S. Interconnected by T1, T3 lines & SONET rings
Call establishment “out-of-band” using packet-switched signaling system (SS7)
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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 AT&T’s 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
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Deregulation of telephony 1960s-70s: gradual de-regulation of AT&T due to
technological advances Terminal equipment could be owned by customers (CPE)
=> explosion in PBXs, fax machines, handsets Modified final judgement (MFJ): breakup of AT&T 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 => retain 800 number
1995: removed price controls on AT&T
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US Telephone Network Structure (after 1984)
Eg: AT&T, Sprint, MCI
Eg: SBC, Verizon, BellSouth
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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 AT&T 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 (AT&T’s 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 => more demand elasticity for
local services
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VoIP Technologies
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IP Telephony Protocols: SIP, RTP
Session Initiation Protocol - SIP Contact “office.com” asking for “bob” Locate Bob’s current phone and ring Bob picks up the ringing phone
Real time Transport Protocol - RTP Send and receive audio packets
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Inside the Endpoint: Data-plane … I.e.after signaling is done… Consists of three components:
UserUser
A/DCodec
A/DCodec
IPGateway
IPGateway
User speaks into microphone, either PC attached, regular analogue phone or IP phone
Device digitizes voice according to certain codecs:
G.711 / G.723.1 / G.729 ...
Voice gets transmitted via RTP over an IP infrastructure
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Internet Multimedia Protocol Stack
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PreambleDestination
AddressData Pad Checksum
SourceAddress
Inter-framegap
Start of framedelimiter
Length orEthertype
12 7 1 6 6 2 0-1500 0-46 4
Ethernet Frame
DestinationAddress
SourceAddress
HeaderChecksum
DataFlags &
Frag OffsetTotal
LengthPacket
IDOptions(if any)
1 1 2 2 2 1 1 2 4 4 0-40 0-1480
IP packet
Version &header length
TOS TTL
Protocol
SourcePort Number
DestinationPort Number
UDP length UDP checksum
2 2 2 2 0-1472
UDP datagram
Version,flags & CC
SequenceNumber
Timestamp
1 1 2 4 4 0-60 0-1460
RTP datagramSynchronization
Source IDPayload
TypeCSRC ID(if any)
Codec Data
Data
Packet Encapsulation
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RTP – Real-time Transport Protocol
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
Version,flags & CC
SequenceNumber
Timestamp
1 1 2 4 4 0-60 0-1460
RTP datagramSynchronization
Source IDPayload
TypeCSRC ID(if any)
Codec Data
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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 don’t implement it
Uses another UDP port (usually RTP’s port + 1)
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SIP
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Signaling: VoIP Camps
ISDN LAN conferencin
g
IP
H.323
I-multimediaWWW
IP
SIP
Call AgentSIP & H.323
IP
“Softswitch” BISDN, AIN
H.xxx, SIP
“any packet”
BICC
Conferencing Industry
Netheads“IP over
Everything”
Circuit switch
engineers “We over
IP”
“Convergence” ITU
standards
Our focus
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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
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SIP
SIP
Audio Codec
G.711
G.723
G.729
Video Codec
H.261
H.263
RTP RTCP
LAN Interface
TCP
IP
UDP
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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
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SIP Addresses Food Chain
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Why is SIP interesting? SIP is IETF’s 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
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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)
4
IP SIP Phones and Adaptors
1
3
Analog phone adaptor
Palmcontrol
2
54
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SIP: Personal Mobility
Users maintain a single externally visible identifier regardless of their network location
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Expand existing PBXs w/ IP phones
Transparently …
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SIP as Event Notification Protocol
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SIP: Presence
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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)
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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)
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SIP-based Architecture
SIP proxy,redirectserver
SQLdatabase
sipd
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
SIPH.323convertor
NetMeetingsip323
H.323
rtspd
SIP/RTSPUnified
messaging
RTSP media server
sipum
Quicktime
RTSP clients
RTSP
SIP conference
server
sipconf
T1/E1 RTP/SIP
Telephone
Cisco 2600 gateway
Telephoneswitch Web based
configuration
Web server
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SIP proxy,redirectserver
SQLdatabase
sipd
e*phone
sipc
Software SIP user agents
Hardware Internet (SIP)
phones
Web based configuration
Web server
Call Bob
Example Call
• Bob signs up for the service from the web as “[email protected]”
• He registers from multiple phones
• Alice tries to reach Bob INVITE ip:[email protected]
• sipd canonicalizes the destination to sip:[email protected]
• sipd rings both e*phone and sipc
• Bob accepts the call from sipc and starts talking
ecse.rpi.edu
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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
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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.
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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.
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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).
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SIP Design Choices
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Proxy server
parliament.uk
Location Server
ge
org
e.w
.bu
sh d
ch
en
ey
@w
h
3. SIP/2.0 200 ok From: sip:dcheney@wh
4. SIP/2.0 100 OK From: sip:[email protected]
1. INVITE sip:[email protected] SIP/2.0 From: sip:[email protected]
5. ACK sip:[email protected] SIP/2.0 From: sip:[email protected]
6. ACK sip:dcheney@wh SIP/2.0 From: sip:[email protected]
2. INVITE sip:dcheney@wh SIP/2.0 From: sip:[email protected]
1 & 5
4
2 & 6
3
Proxy Server
us.gov
dcheney@wh
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parliament.uk
us.gov
Redirect Server
Location Server
ge
org
e.w
.bu
sh d
ch
en
ey
@w
h
2. SIP/2.0 320 Moved temporarily Contact: sip:[email protected]
3. ACK sip:[email protected] From: sip:[email protected]
1. INVITE sip:[email protected] From: sip:[email protected]
6. ACK sip:[email protected] From: sip:[email protected]
4. INVITE sip:[email protected] From: [email protected]
5. SIP/2.0 200 OK To: [email protected]
1 & 3
2
5
4 & 6
Redirect Server
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SIP Call Signaling
Media (UDP)RTP StreamRTCP Stream
SIPEndpoint
SIP + SDP (TCP or UDP)
Invite
RTP Stream
SIPGateway
Assumes Endpoints(Clients) know each other’s IP addresses
Signaling Plane
BearerPlane
200 OK
Ack
180 Ringing
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PSTN to IP Call
PBXPSTN
External T1/CAS
Regular phone(internal)
Call 93971341
SIP server
sipd
Ethernet
3
SQLdatabase
4 7134 => bob
sipc
5
Bob’s phone
GatewayInternal T1/CAS(Ext:7130-7139)
Call 71342
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IP to PSTN Call
Gateway(10.0.2.3)
3
SQLdatabase
2Use sip:[email protected]
Ethernet
SIP server
sipdsipc
1Bob calls 5551212
PSTN
External T1/CAS
Call 55512125
5551212
PBX
Internal T1/CASCall 85551212 4
Regular phone(internal, 7054)
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Traditional voice mail system
Alice939-7063
Bob853-8119
Dial 853-8119
Phone is ringing
.. The person is not available nowplease leave a message ...
... Your voice message ...
Disconnect
Bob can listen to his voice mails by dialing some number.
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SIP-based Voicemail Architecture
INVITE [email protected]
Alice
phone1.office.com
Bob
Alice calls [email protected] through SIP proxy.SIP proxy forks the request to Bob’s phone as well as to a voicemail server.
vm.office.com
The voice mail server registers with the SIP proxy, sipd
INVITE [email protected]
INVITE [email protected]
REGISTER [email protected]
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Voicemail Architecture
v-mail
rtspd
Alice
vm.office.com;
200 OK
200 OK
CANCEL
SETUP
RTP/RTCP
phone1.office.com;
Bob
After 10 seconds vm contacts the RTSP server for recording.
vm accepts the call.Sipd cancels the other branch and ......accepts the call from Alice.Now user message gets recorded
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IETF SIP Architecture Tour: RoundupRegistrar & Proxy or Redirect Server
Registrar & Proxy or Redirect Server
*Gateway*Gateway
*User Agent*User Agent*User Agent*User Agent*User Agent*User Agent
Media streams: RTP/RTCP (G.911, G.723.1, … )
PSTN,ISDN,ATM,etc
*Endpoints
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IETF SIP Architecture Tour: RoundupRegistrar & Proxy or Redirect Server
Registrar & Proxy or Redirect Server
*Gateway*Gateway
*User Agent*User Agent*User Agent*User Agent*User Agent*User Agent
Media streams: RTP/RTCP (G.911, G.723.1, … )
PSTN,ISDN,ATM,etc
*Endpoints
Interface to non-IP or H.323
networks
Interface to non-IP or H.323
networks
End-user devicesand network proxies
End-user devicesand network proxies
Conferencing does not need another
box (MCU)
Conferencing does not need another
box (MCU)
System Management• admission control• address
translation/forwarding• Firewall bypassing
System Management• admission control• address
translation/forwarding• Firewall bypassing
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IETF SIP Architecture Tour: RoundupRegistrar & Proxy or Redirect Server
Registrar & Proxy or Redirect Server
*Gateway*Gateway
*User Agent*User Agent*User Agent*User Agent*User Agent*User Agent
Media streams: RTP/RTCP (G.911, G.723.1, … )
PSTN,ISDN,ATM,etc
*Endpoints
Components of the SIP protocol suite:•SIP = almost all signaling, optional services, etc.•SDP = negotiation/capabilities •DNS = address translation•RSVP = QoS bandwidth guarantee
Components of the SIP protocol suite:•SIP = almost all signaling, optional services, etc.•SDP = negotiation/capabilities •DNS = address translation•RSVP = QoS bandwidth guarantee
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SDP: Session Description Protocol Not really a protocol – describes data carried by other
protocols Used by SAP, SIP, RTSP, H.332, PINT. Eg:
v=0o=g.bell 877283459 877283519 IN IP4 132.151.1.19s=Come here, Watson!u=http://[email protected]=IN IP4 132.151.1.19b=CT:64t=3086272736 0k=clear:manhole coverm=audio 3456 RTP/AVP 96a=rtpmap:96 VDVI/8000/1m=video 3458 RTP/AVP 31m=application 32416 udp wb
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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
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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
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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
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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 (TLS+TCP, TCP, UDP, SCTP)
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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 SIP’s 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.
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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 sip:[email protected];comp=sigcomp Via: SIP/2.0/UDP
server1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp SIGCOMP is the compression protocol
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Device Configuration
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SIP Scaling Issues
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SIP Scaling (contd)
SIP Load Characteristics:
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H.323
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SIP vs H.323 vs Megaco
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H.323 vs SIP
IP and lower layers
TCP UDPTPKT
Q.931 H.245 RAS RTCPRTP
Codecs
Terminal Control/Devices
Transport Layer
SIP SDPRTP
CodecsRTCP
Terminal Control/Devices
Typical UserAgent Protocol stack for Internet
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SIP versus H.323
• Complex, monolithic design• Difficult to extend & update• Based on H.320 conferencing and
ISDN Q.931 legacy (“Bell headed”) • Powerful for video-conferencing
• Modular, simplistic design• Easily extended & updated• Based on Web principals (“Internet-
friendly”)• Readily extensible beyond telephony
Properties
• H.450.x series provides minimal feature set only, and not implemented by many
• Options and versions cause interop problems
• Slow moving
• Few real end-device features standard, and not implemented by many
• Many options for advanced telephony features
• Good velocity
Stds Status(end device)
•ITU-T SG-16 •IETF SIPStds Body
• Established now, primarily system level• Few H.323-based telephones• End-user primarily driven by Microsoft
(NetMeeting), Siemens, Intel
• Rapidly growing industry momentum, at system and device level
• Growing interest in SIP-phones and soft clients
IndustryAcceptance
H.323 SIP
H.323 and SIP are direct competitors in peer-level call control space
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SIP-H.323: Interworking ProblemsEg: Call setup translation
Q.931 SETUP
Q.931 CONNECT
INVITE
200 OK
ACK
Terminal Capabilities
Terminal Capabilities
Open Logical Channel
Open Logical Channel
H.323 SIP
Destination address ([email protected])
Media capabilities (audio/video)
Media transport address (RTP/RTCP receive)
• H.323: Multi-stage dialing
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H.323 Standard Series
System Control
H.245 Control
H.225 Call Setup
RAS Gatekeeper
Audio Codec
G.711
G.723
G.729
Video Codec
H.261
H.263
RTP RTCP
LAN Interface
TCP
IP
UDP
Data Interface
T.120
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Internet Telephony Protocols: H.323
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H.323 (contd) Terminals, Gateways, Gatekeepers, and Multipoint
Control Units (MCUs)
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H.323 Model - Gatekeeper Routed Call
Call S
etup
/Sign
aling
Voice Channel
Gatekeeper
Endpoint
Gateway
Call Setup/SignalingCall
Con
trol
RAS Call Control
RAS
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H.323 Model - Gatekeeper Direct Call
Call Setup/Signaling
Call Control
Voice Channel
RAS RAS
Gatekeeper
EndpointGateway
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H.323 Call Signaling
Media (UDP)RTP StreamRTCP Stream
H.323Endpoint
H.245 (TCP)Open Logical Channel
H.225 (TCP)(Q.931)
Setup
Connect
Open Logical Channel & Acknowledge
RTP Stream
H.323Gateway
H.323v1 (5/96) - 7 or 8 Round TripsH.323v2 Fast Start (2/98) - 2 Round Trips
Assumes Endpoints(Clients) know each other’s IP addresses
Signaling Plane
BearerPlane
Alerting
Terminal Capability Set
Terminal Capability Set & Acknowledge
Terminal Capability Set Acknowledge
Open Logical Channel Acknowledge
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ITU-T H.323 Architecture TourGate Keeper
(GK)
Gate Keeper(GK)
*Gateway (GW)*Gateway (GW)
*Terminal*Terminal
*Multipoint ControlUnit (MCU)
*Multipoint ControlUnit (MCU)
MultipointController
(MC)
MultipointProcessor
(MP)*Terminal*Terminal*Terminal*Terminal
Media streams: RTP/RTCP (G.911, G.723.1, … )
PSTN,ISDN,ATM,etc
*Endpoints
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ITU-T H.323 Architecture TourGate Keeper
(GK)
Gate Keeper(GK)
*Gateway (GW)*Gateway (GW)
*Terminal*Terminal
*Multipoint ControlUnit (MCU)
*Multipoint ControlUnit (MCU)
MultipointController
(MC)
MultipointProcessor
(MP)*Terminal*Terminal*Terminal*Terminal
Media streams: RTP/RTCP (G.911, G.723.1, … )
PSTN,ISDN,ATM,etc
*Endpoints
Interface to non-IP networks
Interface to non-IP networks
End-user devicesand network proxies
End-user devicesand network proxies
ConferencingConferencing
System Management• zone management• b/w management &
admission control• address translation• centralized control
(“gatekeeper control mode”)
System Management• zone management• b/w management &
admission control• address translation• centralized control
(“gatekeeper control mode”)
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ITU-T H.323 Architecture TourGate Keeper
(GK)
Gate Keeper(GK)
*Gateway (GW)*Gateway (GW)
*Terminal*Terminal
*Multipoint ControlUnit (MCU)
*Multipoint ControlUnit (MCU)
MultipointController
(MC)
MultipointProcessor
(MP)*Terminal*Terminal*Terminal*Terminal
Media streams: RTP/RTCP (G.911, G.723.1, … )
PSTN,ISDN,ATM,etc
*Endpoints
Components of the H.323 protocol suite:•Q.931 = ISDN call signalling•H.225.0 = RAS (registration/admissions/status) gatekeeping functions
+ Call signalling channel (CS), contains Q.931•H.245 = Control channel (CC), negotiation/capabilities, logical signalling,
maintenance •H.450.x = Supplementary services (SS), transfer, hold, park, msg wait, … incomplete!
Components of the H.323 protocol suite:•Q.931 = ISDN call signalling•H.225.0 = RAS (registration/admissions/status) gatekeeping functions
+ Call signalling channel (CS), contains Q.931•H.245 = Control channel (CC), negotiation/capabilities, logical signalling,
maintenance •H.450.x = Supplementary services (SS), transfer, hold, park, msg wait, … incomplete!
H.225.0 RAS
H.225.0 CSH.245 CCH.450.x SS
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Gatekeeper132.177.120.5
223-274910.0.0.5
3. Connect
4. Connect
1. Setup called: 5551234 caller: 9642749::10.0.0.5
5. TCS media: G.711/30ms, G.729/30ms
2. Setup called: 5551234::192.168.0.3 caller: 9642749
1, 5, 9, 13
4, 8, 12, 16
2, 6, 10, 14
3, 7, 11, 15
Gatekeeper Routed Call
Atlanta Zone (404)
223-4211192.168.0.3
6. TCS media: G.711/30ms, G.729/30ms
7. TCS media: G.729/20ms, G.723
8. TCS media: G.729/20ms, G.723
9. Open Channel G.729/30ms, 10.0.0.5:6400
10. Open Channel G.729/30ms, 10.0.0.5:6400
11. Open Channel G.729/20ms, 192.168.0.3:2300
12. Open Channel G.729/20ms, 192.168.0.3:2300
13. ACK 14. ACK 15. ACK 16. ACK
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Gatekeeper132.177.120.5
223-274910.0.0.5
3. Setup called: 5551234 caller: 9642749::10.0.0.5
4. Connect
1. ARQ called: 5551234 caller: 9642749::10.0.0.5
5. TCS media: G.711/30ms, G.729/30ms
2. ACF called: 5551234::192.168.0.3
1
4, 6, 8, 10
2
3, 5, 7, 9
Gatekeeper Direct Call
Atlanta Zone (404)
223-4211192.168.0.3
6. TCS media: G.729/20ms, G.723
7. Open Channel G.729/30ms, 10.0.0.5:6400
8. Open Channel G.729/20ms, 192.168.0.3:2300
9. ACK
10. ACK
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MEGACO/H.248, Softswitch Concepts
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Master/Slave vs. Peer Comparison
•Lowest cost end device •Higher cost end deviceCost
•Lower performance “local” services•Sometimes higher
performance distributed services (e.g.. call control)
•Higher performance local services•High performance User
Interface
Performance
Feature deployment
•Update servers only•Services can come and go dynamically
•Update / download all end devices in network (yikes!)•Features more static per-device
Master/Slave (Thin Client) Peer (Thick Client)•Simple/dumb slave end device•Stimulus control, proxy in
network
•Smart/complex end device•Functional control, peer
interaction
Operation
Feature development
•Generic development tools•Shorter time to market for new features on a range of end devices•End device does not “get out of date” as quickly
•Device-specific development•Possibly shorter time to market
for new features on specific devices
•End device may need hardware upgrade over time
•MEGACO/H.248, MGCP •H.323, SIPProtocols
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Megaco/H.248
Megaco
Audio Codec
G.711
G.723
G.729
Video Codec
H.261
H.263
RTP RTCP
LAN Interface
TCP
IP
UDP
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Megaco/H.248 – Convoluted History
DSM-CCDSM-CC
DiameterDiameter
SGCPSGCP
IPDCIPDC
MGCP(proposal)
MGCP(proposal)
MDCP(proposal)
MDCP(proposal)
Megaco ProtocolMegaco Protocol
PacketCable NCS
PacketCable NCS
Megaco/H.248Megaco/H.248
Agreement reached between ITU SG16 and IETF Megaco to work together to create one standard (Summer 99)
ITU: H.GCPITU: H.GCP
MGCP proposal
PacketCable Network-based Call Signaling (NCS) based on earlier version of MGCP (March 99)
Megaco Protocol stream created, true consensus (March 99)
ITU SG-16 initiates gateway control project, H.GCP starting
from MDCP (May 99)
I-RFC 2705I-RFC 2705
MGCP released as Informational RFC (Oct 99)
WORLDSTANDARD
Industry Defacto
Std.
Non-Standard
Not fully accepted by Megaco WG, diverged (Spring 99)
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Megaco Vs MGCP
Call Model Termination +Context +Topology P2P Single Media Single Media Conferencing P2P Multimedia Multimedia ConferencingTerminations Physical & Ephemeral & Muxing Template
Megaco/H.248
Event Packages (MGCP)Media Session Description SDPProtocol Encoding TextTransport UDP
MGCP
Command Grouping TransactionEvents Event BufferingEvent Packages (MGCP Packages + Additional Packages) National VariantsMedia Session Description SDP + H.245
Call Model Termination + Connection P2P Single Media Single Media ConferencingTerminations Physical & EphemeralCommand Grouping Ad hoc EmbeddingEvent Quarantine
Protocol Encoding Binary & TextTransport TCP + UDP +SCTPSecurity Authentication HeaderMGC Backup
Bold entries indicate additional features in Megaco vs. MGCP
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Endpoint
(e.g.. H.323 Gateway,
Terminal, MCU)
Gateway Function
(e.g.. H.323 Gateway,
Terminal, MCU)
Call Agent
Media Gateway Controller
Signalling Gateway
PSTN,
ATM,
etc
trunks
lines
SS7 etc
Sigtran
AnalogMedia Gateway
PSTN trunkingMedia Gateway
PSTN lineMedia Gateway
IP PhoneMedia Gateway
Megaco Scope
Megaco Architecture Whirlwind Tour
Media Gateway Control Layer (MGC)• Contains all call control intelligence• Implements call level features (forward,
transfer, conference, hold, …)
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
Signalling Gateway Layer (SG)• Interface to SS7 signalling etc• Not in Megaco scope (IETF Sigtran)
Media Gateway Control Protocol• Master / slave control of MGs by MGCs
–Connection control–Device control and configuration
• Orthogonal to call control protocols
Megaco Protocol
Call control (e.g.. H.323, SIP…)
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Framework for H248/Megaco Protocol
IP (or ATM) Network
PBX/CO
PBX/CO
Media GW Controller
Media Gateway
Media Gateway• Connection and device control• No call processing, no call model• Service-independent• Cost effective
Devicecontrol
Devicecontrol
IP PhoneMedia
Gateway
Telephone/ResidentialMedia Gateway
PSTN trunkingMedia Gateway
PSTN lineMedia Gateway
Media GW Controller• Call processing and Service logic• Call routing• Inter-peer entity communication via
call control protocols (e.g. H.323, SIP, etc)
PBXMedia
Gateway
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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 GW Controller
Media Gateways
Megaco/H.248
PSTN TrunkingMedia Gateway
PSTN LineMedia Gateway
Telephone/ResidentialMedia Gateway
Cable ModemMedia Gateway
Virtual SwitchSS7 Signalling
Gateway
Sigtrans
H.323H.323
Device
SIP
SIPDeviceRTP
RTP
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Megaco call in action (optional)
ServiceChange: Restart
Reply: ServiceChangeReply: ServiceChange
Modify: Look for Off-Hook
MG1 MG2
Dial Tone,User Dials
Powered On
Powered On
ServiceChange: Restart
Modify: Look for Off-Hook
Ready ReadyReply: Modify Reply: Modify
MGC
Notify: Off-HookOff-HookReply: Notify
Modify: Dial Tone, Digit Map
Reply: Modify
Notify: number “19782886160”
Reply: Notify
Add: TDM to RTP, what codecs?
Reply: Add, codec G.729
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Megaco call in action (continued)
Reply: Add
Add: TDM to RTP, ring phone
MG1 MG2
Open RTP Open RTPActive Call/End of Invite Request
Phone Rings
MGC
Modify: ip of MG2, ringback
Reply: ModifyHears Ring Off-HookNotify: Off-hook
Reply: Notify
Modify: stop ringStops Ring
Reply: ModifyModify: stop ringback, fullduplexReply: Modify
On-HookNotify: On-hookReply: Notify
Subtract:TDM and RTP
Reply: Subtract
Subtract: TDM and RTP
Reply: Subtract
Disconnect
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Megaco/H.248 IP Phone Control
H.323MGC
Med
ia, L
CD, Sof
tkey C
ontro
l Media, LCD, Softkey Control
IP PhoneMedia Gateway
IP PhoneMedia Gateway
Voice (RTP)
Voice (RTP)
Voice
(RTP
)Voice (RTP)
In theory the RTP stream should go direct phone<->GW, but many today tandem through the MGC
In theory the RTP stream should go direct phone<->GW, but many today tandem through the MGCV
oic
e (R
TP
)
H.323 GWCisco’s Skinny,Nortel’s UNIStim,etc., are very similar protocols but they’re not interoperable
Cisco’s Skinny,Nortel’s UNIStim,etc., are very similar protocols but they’re not interoperable
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Vendor Support for Standards
Source: Network World and Mier Communications - August, 2001
VoIP Protocol Support
30
64
54
56
81
66
32
57
77
22
11
30
42
30
40
17
26
73
0 10 20 30 40 50 60 70 80 90
Other
H.248 (Megaco)
MGCP (latest spec)
MGCP (orig. RFC2705)
SIP (Latest spec)
SIP (orig. RFC 2543)
H.323 other versions
H.323 V2
H.323 V1
Percentage of Vendors currentlly supporting the protocol
Percentage of vendors planning to add support within the next year
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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
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MGCP/H.248/Megaco
Media Gateway Controller(MGC)
Media Gateway
Media Gateway Controller(MGC)
SIP
Media GatewaySignaling Gateway Signaling Gateway
MGCP
Distributed entities acting in co-ordination
Connect to varietyof networks, home usersand other media receptorslike H.323 terminals etc
Interface tovariety of signaling mechanisms
User A
Separate signaling and voice planes, but
user unaware of it
Master/Slave
For examples of gateways see RFC 3435
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Softswitch: Motivation
Class 5switch
Class 4switch
Class 5switch
UsersUsers
PSTN
Packet networks
Data
Voice
H.323 gateway
ISDN Switch
Class-4/5 switches bulky, expensive. Incentive to switch to cheaper easily managed IP
Initial gateway between PSTN and Internet was H.323. Gateway did signaling, call control, translation in one box. Not scalable.
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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
• It’s an Architecture, NOT a box
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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
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Softswitch: What’s 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
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Softswitch Architecture
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
• Distributed functionality
• Open platforms
• Open interfaces enable new services
• Leverages the intelligence of endpoints
• Media agnostic
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Softswitch - Media Gateway ControllerAn SS7 Enabled Media Gateway Controller integrates the functionality of new applications with the large installed based of legacy systems.
• Multiple controllers can collaborate on a single call
• May be distributed across the globe
• May or may not be collocated with SS7 Signaling Gateway
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
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Softswitch - Media Gateway Controller Functions
• Connections (call setup and teardown)
• Events (detection and processing)
• Device management (gateway startup, shutdown, alerts)
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
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Softswitch - Media GatewaysMedia Gateways provide interaction between audio in the network and software controlled applications
• 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
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
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MGC and MG RolesMedia Gateway Controller
MGC’s allow intelligence to be distributed in the network
Basic call routing functions
Synchronization of Media Gateways
Protocol Conversion
Media Gateway
MG’s are purpose built specialist devices
Trunking gateways VoATM gateways Access gateways Circuit switches Network Access Servers
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Softswitch - Signaling GatewaySignaling Gateways provide interaction between the SS7 network and Media Gateway Controllers.
• Convert SS7 to IP packets
• Convert IP to SS7 packets
• Signaling transport (SS7, SIP-T, Q.931…)
• Extremely secure
• Extremely fault tolerant
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
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Softswitch – Application ServerApplication Servers(AS) provide the new services that are the real “value-add” for Softswitches.
• Many core features are part of the MGC
• Allows new features to be developed by third parties
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
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Softswitch – Application ServerApplication Servers(AS) Can be broken apart and distributed in the network
Feature Server
Policy Server
Directory Server
Media Server Management Server
LDAP
Corba
SIP,Parlay,JAIN
Connectivity Server
SIP
COPSNetwork Elements
Corba
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Softswitch Architecture – The protocolsApplication
Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
SIP, Parlay, Jain
Sigtran w/SCTP
H.248,MGCP
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Softswitch Architecture – Interdomain protocols
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
SIP, Parlay, Jain
Sigtran
H.248,MGCP
Application Server
Media Gateway
Signaling Gateway
Media GatewayController
PSTN/ End users
SIP-T,BICC
RTP
Application specific
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SIP vs MEGACO: Summary
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SIP vs MEGACO (contd)
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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)
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Speech Coding and Speech Coders for VoIP
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Taxonomy of Speech CodersSpeech Coders
Waveform Coders Source Coders
Time Domain: PCM, ADPCM
Frequency Domain: e.g. Sub-band coder,Adaptive transform coder
Linear Predictive Coder
Vocoder
Waveform coders: attempts to preserve the signal waveform not speech specific (I.e. general A-to-D conv)
PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbpsVocoders:
Analyse speech, extract and transmit model parameters Use model parameters to synthesize speech LPC-10: 2.4 kbps
Hybrids: Combine best of both… Eg: CELP (used in GSM)
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Speech Quality of Various Coders
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Speech Quality (Contd)
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Actual Bandwidth Used
Framesizein ms
PacketIn bytes
+ RTP+UDP+IPin bytes
LANframe inbytes
T-LANkbps
WANkbps
G.711(64 kbps)
102030
80160240
120200280
146226306
116.890.481.5
96.080.074.6
G.729A/G.729( 8 kbps)
102030
102030
506070
768696
60.834.425.6
40.024.018.6
G.723.1(5.3 kbps)
30 20 60 86 22.9 16.0
G.723.1(6.3 kbps)
30 24 64 90 24.0 17.0
Note: (1) 26-bytes Ethernet overhead was removed for WAN calculation. (2) No backbone protocol overhead was used for WAN bandwidth. (3) This is per voice direction, so multiply by 2 if on a shared (half-duplex) media
(4) No Ethernet Interframe Gap was included (another 12 bytes)
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Applications of Speech Coding
Telephony, PBX Wireless/Cellular Telephony Internet Telephony Speech Storage (Automated call-centers) High-Fidelity recordings/voice Speech Analysis/Synthesis Text-to-speech (machine generated speech)
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Pulse Amplitude Modulation (PAM)
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Pulse Code Modulation (PCM)
* PCM = PAM + quantization
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Companded PCM
•Small quantization intervals to small samples and large intervals for large samples• Excellent quality for BOTH voice and data• Moderate data rate (64 kbps)• Moderate cost: used in T1 lines etc
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How it works for T1 Lines
• Companding blocks are shared by all 16 channels
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Recall: Taxonomy of Speech CodersSpeech Coders
Waveform Coders Source Coders
Time Domain: PCM, ADPCM
Frequency Domain: e.g. Sub-band coder,Adaptive transform coder
Linear Predictive Coder
Vocoder
Waveform coders: attempts to preserve the signal waveform not speech specific.
PCM 64 kbps, ADPCM 32 kpbs, CVSDM 32 kbpsVocoders:
Analyse speech, extract and transmit model parameters Use model parameters to synthesize speech LPC-10: 2.4 kbps
Hybrids: Combine best of both… Eg: CELP
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Vocoders
Encode only perceptually important aspects of speech w/ fewer bits than waveform coders: eg: power spectrum vs time-domain accuracy
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LPC Analysis/Synthesis
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Speech Generation in LPC
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CELP Encoder
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Example: GSM Digital Speech Coding
PCM: 64kbps too wasteful for wireless
Regular Pulse Excited -- Linear Predictive Coder (RPE--LPC) with a Long Term Predictor loop.
Subjective speech quality and complexity (related to cost, processing delay, and power)
Information from previous samples used to predict the current sample: linear function.
The coefficients, plus an encoded form of the residual (predicted - actual sample), represent the signal.
20 millisecond samples: each encoded as 260 bits =>13 kbps (Full-Rate coding).
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Standard Algorithm Bit Rate (Kbit/s) Codec Induced Delay (msecs)
Resultant Voice Quality
G.711 PCM 56, 64 <<1 Excellent G.723.1 MPE/ACELP 5.3, 6.3 67-97 Fair(5.3), Good(6.3) G.728 LD-CELP 16 <<2 Good G.729 CS-ACELP 8 25-35 Good G.722 Sub-band
ADPCM 64 5-10 Good-Excellent (it’s
wideband) G.726 ADPCM 16, 24, 32, 40 <<1 Fair(24), Good(40) GSM-EF ACELP 12.2 40 Good
Codecs: Quality Measures
Only G.711, G.723.1, and G.729 are popular (because they are mandatory for several specs)
G.711 is the best (obviously), but G.729 isn’t much worse G.723.1 is HORRIBLE
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PreambleDestination
AddressData Pad Checksum
SourceAddress
Inter-framegap
Start of framedelimiter
Length orEthertype
12 7 1 6 6 2 0-1500 0-46 4
Ethernet Frame
DestinationAddress
SourceAddress
HeaderChecksum
DataFlags &
Frag OffsetTotal
LengthPacket
IDOptions(if any)
1 1 2 2 2 1 1 2 4 4 0-40 0-1480
IP packet
Version &header length
TOS TTL
Protocol
SourcePort Number
DestinationPort Number
UDP length UDP checksum
2 2 2 2 0-1472
UDP datagram
Version,flags & CC
SequenceNumber
Timestamp
1 1 2 4 4 0-60 0-1460
RTP datagramSynchronization
Source IDPayload
TypeCSRC ID(if any)
Codec Data
Data
Packet Encapsulation
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42668 120IP into Ethernet
IP PayloadPreamble
Destination
Source
Type CRC.
80 byte voice bundles
RTP Frame 12RTP Header Voice Payload
80
2222 80UDP Datagram
Voice Payload
Destination
Source
LengthChecksum
Destination
12
RTP Header
1 12 2120IP into Frame Relay Flag
FlagAddress Frame Check
IP Payload
4412 80IP Packet Header
SourceVoice PayloadUDP Header RTP Header
8 12
5 48 5 24+IP into ATMIP Payload
5 48 +IP Payload IP Payload
Header Header Header
16
Padding Trailer
8
G.711 (10ms) Clear Channel Voice
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42668 70IP into Ethernet
IP PayloadPreamble
Destination
Source
Type CRC.
30 byte voice bundles
RTP Frame 12RTP Header Voice Payload
30
2222 30UDP Datagram
Voice Payload
Destination
Source
LengthChecksum
Destination
12
RTP Header
1 12 270IP into Frame Relay Flag
FlagAddress Frame Check
IP Payload
4412 30IP Packet Header
SourceVoice PayloadUDP Header RTP Header
8 12
5 48 5 22IP into ATMIP Payload
+IP Payload
Header Header
18
Padding Trailer
8
G.729 (30ms) Clear Channel Voice
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42668 60IP into Ethernet
IP PayloadPreamble
Destination
Source
Type CRC.
20 byte voice bundles
RTP Frame 12RTP Header Voice Payload
20
2222 20UDP Datagram
Voice Payload
Destination
Source
LengthChecksum
Destination
12
RTP Header
1 12 260IP into Frame Relay Flag
FlagAddress Frame Check
IP Payload
4412 20IP Packet Header
SourceVoice PayloadUDP Header RTP Header
8 12
5 48 5 12IP into ATMIP Payload
+IP Payload
Header Header
28
Padding Trailer
8
G.729 (20ms) Clear Channel Voice
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20-24 byte voice bundles
RTP Frame 12RTP Header Voice Payload
20-24
42668 60-64IP into Ethernet
IP PayloadPreamble
Destination
Source
Type CRC.
2222 20-24UDP Datagram
Voice Payload
Destination
Source
LengthChecksum
Destination
12
RTP Header
1 12 260-64IP into Frame Relay Flag
FlagAddress Frame Check
IP Payload
4412 20-24IP Packet Header
SourceVoice PayloadUDP Header RTP Header
8 12
5 48 5 12-16IP into ATMIP Payload
+IP Payload
Header Header
28-24
Padding Trailer
8
G.723.1 (30ms) Clear Channel Voice
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Coding Technology Side-effects
Coded VoIP is NOT the same as a telephone line (I.e. it is not a content-neutral “carrier”): Without special support, you cannot send “fax” or “modem
traffic” over VoIP The “carrier” is now IP (or some data-transport protocol
like frame-relay or ATM) The same is true for 3G or GSM telephony Why? Voice is encoded and the encoding works only for
voice! (it is no longer a 64 kbps bit stream) Fax support: Fax Passthru, T.38 fax Relay
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Voice Quality: Loss Tolerance Voice codecs are unevenly tolerant of packet loss,
but loss above 2 to 5 percent will have a perceptible effect on quality.
Losses also associated with higher jitter 1-way delay > 150 milliseconds, => trouble Jitter buffer (major component of delay budget) Capacity reservations & priority for key packets:
setup through RSVP Priority: using TOS bits: 8 levels of precedence
Carrier networks use some combination of: MPLS (traffic engineering, stable routing) and Diff-serv (expedited forwarding) to provide
superior service for VoIP
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VoIP QoS Myths Packet voice=> voice could take multiple paths or failover.
But it usually does not…
VoIP is sensitive to routing failures or congestion in paths OSPF and BGP convergence times too bad for VoIP:
SONET and (now) MPLS much better
However, FEC packets for VoIP can be sent on a separate path or on the same path: hedge against performance fluctuations (eg: congestion) on
the primary path, but limited hedge against failure of the primary path.
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Voice codecs: Summary G.711
uncompressed PCM audio stream 8ks/s of 8 bit values = 64kbps packet “sizes” = 10, 20, 30 and 60ms
G.722 - Wideband (7kHz) G.726
ADPCM - 10,20,30,60ms - 32kbps G.723.1
MLQ - 30ms - 5.3 or 6.3kbps Silence suppression
G.729 CS-ACELP - 10, 20, 30ms - 8kbps Annex B adds silence suppression
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Recap: Speech Quality of Various Coders
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Miscl: Other standards, ENUM, E-911, Presence etc
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Sigtrans (Signaling Transport) Signalling transport protocol and adaptation layers for SG to
MGC communication, and for SG to SG communication Signalling Gateways can be stand-alone or co-located with an
MGC
Media GW Controller
Trunk Gateway
Megaco/H.248
Virtual Switch
Sigtrans SIP, H.323
Media Gateway
Signalling Gateway
PBX
D-channel
B-channels
PRI
Virtual Switch
RTP
Sig
tran
s
Signaling Gateway
Signaling Gateway
SS7
CO
Sigtrans
Sig
tran
s
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SCTP (Stream Control Transmission Protocol)
Sigtrans needs to carry SS7 Needed a reliable transport mechanism (like TCP) without the overhead
of a connection-oriented protocol SCTP created: like UDP, but with acknowledgment, fragmentation, and
congestion-avoidance This has much broader use than just carrying SS7: it’s being looked at
for SIP, RTP, T.38, and more...
6 - Presentation5 - Session User Adaptation Modules 4 - Transport SCTP3 - Network IP2 - Link MLPPP / FR / ATM1 - Physical Ethernet / SONET/Serial
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(1) SS7 Signaling Using IP Transport
Applications
MTP2
IP
SCTP
SSP
STP
MTP2
MTP3
SCCP
TCAP
ISUP
Applications
SSP
IP
SCTP
The IETF M2UAMTP2-User Adaptation Layer
from the Sigtran WG
M2UA
MTP3*
M2UA
MTP3
SCCP
TCAP
ISUP
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(2) SS7 / IP Interworking
CallProcessingApplication
MTP2IP
SCTP
SSP MGC
SS7 SG
M3UA
Nodal Inter-working Function
CallProcessingApplication
MTP3
MTP2
MTP3
ISUP
IP
SCTP
M3UA
ISUP
The IETF M3UAMTP3-User Adaptation Layer
from the Sigtran WG
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BICC (Bearer Independent Call Control) Offers a migration path from SS7/TDM to packet-based
voice Defines Interface Serving Node for Bearer, Bearer Control,
and Call Serving Functions Specifies Transit Serving Nodes to change bearer types,
and Gateway Serving Node to transit operators
PSTN PSTN
Class 4 Switch Class 4 Switch
Data Network
BICC ISN BICC ISN
BICCISUP ISUP
TDMTDM
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VPIM (Voice Profile for Internet Mail) Uses SMTP to send/receive voice/faxmail messages Attaches messages as wav/mpeg/tiff files in MIME Useful for transferring across voicemail systems Adds more useful info: vcard, signature, multiple addresses POP3 still used to download voicemail to your favorite email
client (Outlook, Eudora, Pine, etc.)
PBXUnified
Messaging System
Unified Messaging
System
SIP/H.323
SIPDevice
VPIMEmail
Browser
POP3
Plain Phone
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TRIP – Telephony Routing over IP
TRIP is a protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the routes to those destinations.
Can serve as a routing protocol for any signaling protocol TRIP is used to distribute telephony routing information
between telephony administrative domains. TRIP is essentially BGP for phone numbers and the
protocol is actually based on BGP-4
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Proxy server
parliament.uk
Location Server
ge
org
e.w
.bu
sh d
ch
en
ey
@w
h
3. SIP/2.0 200 ok From: sip:dcheney@wh
4. SIP/2.0 100 OK From: sip:[email protected]
1. INVITE sip:[email protected] SIP/2.0 From: sip:[email protected]
5. ACK sip:[email protected] SIP/2.0 From: sip:[email protected]
6. ACK sip:dcheney@wh SIP/2.0 From: sip:[email protected]
2. INVITE sip:dcheney@wh SIP/2.0 From: sip:[email protected]
1 & 5
4
2 & 6
3
Midcom (Middlebox Communication)
us.gov
dcheney@wh
3.5 Midcom Protocol
Firewall
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Mediation and Billing
Current State
Non real time Non-scalable Limited functionality No revenue assurance capabilities Proprietary CDR formats No OSS functionality (fraud, churn, etc.) Mainly stand alone systems (no integration with the
legacy systems)
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Call Detail Records
• To be able to run reports and bill, Call Detail Records (CDRs) must be recorded for each call:
With VoIP far more detail is necessary: Packets transmitted Packets lost Jitter Delay Call Control / Gateway used Codec used …
Time Reason From To Duration
Details
16:45 Call req. 5551212 6663434 01:45 Normal disc.
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190
Mediation and Billing Requirements Complete call details including
Call descriptors caller ID, called #, time, length, disconnect reason, QoS requested, etc.,
Complete network QoS information (dropped packets, trunk failure, etc.)
Complete application level QoS (dropped frames, disconnect reason, CODEC type, etc.)
Carrier-grade solution Scalable Large number of calls/sec Cover large, distributed networks
Real Time Revenue Assurance
99.999% accuracy Audit capabilities Highly available
Support of standards Integration with other OSS/BSS systems (fraud, churn, etc)
Fault tolerantLocal cache Roll back
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191
IPDR – IP Data Records
The purpose of the IPDR initiative is to define the essential elements of data exchange between network elements, operation support systems and business support systems. Specific goals include:
Define an open, flexible record format (the IPDR record) for exchanging usage information.
Define essential parameters for any IP transaction. Provide an extension mechanism so network
elements and support systems exchange optional usage metrics for a particular service.
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ENUM vs DNS DNS (or internet) names: interpreted right to left:
Eg: www.rpi.edu Telephone numbers: interpreted left to right:
Eg: +1 518 276 8979 ENUM: (RFC 3761)
telephone numbers written DNS-style, Rooted at the domain e164.arpa. So, 1.212.543.6789 becomes 9.8.7.6.3.4.5.2.1.2.1.e164.arpa. When queried, DNS can return an IP address for the telephone
number, or it can return a rule for re-formatting the original number For example, rules can be returned to rewrite 1.212.543.6789 as
sip:[email protected], sip:[email protected].
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193
Continuity of Telephone Svcs in VoIP
A number of basic features remain same: Phone looks and behaves like a phone DTMF (touch-tone) features: mid-call signaling E.911 will provide 911 location services Bearer (“data-plane”) is separated from signaling
(“control-plane”) and is handled differently But, unlike telephony, it is multiplexed on the
same network Interfaces smoothly with internet applications: IM,
Web, email…
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194
E911 - Requirements
911 Services
Power stays on when building power fails
Need callers phone number and location
Services must be modified during a 911 callDisable call-waitingDisable three-party callsCaller cannot hangup and place another call
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195
E911 – VoIP Enhancements
VoIP has the potential of enhancing E911 functionality Multimedia communication
Audio – emulate existing servicesVideo – images and/or biometrics to/from emergency
techniciansText – for hearing impaired
Call setup could contain medical backgroundCan be locally maintained, does not a master database
Calls can easily be forwarded or transferredFast call setup times
PSAP could easily be deployed or relocated anywhere Internet access is available.
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196
E911 – Using DNS to convey location Based on network device name
pigface 192.168.200.20
GL S3.US.95401.4500 “110 Stony Point Rd.,Santa Rosa CA” Based on Geographic location (longitude/latitude)
pigface 192.168.200.20
GPOS -38.43954 122.72821 10.0 Binary (includes precision indicator)
pigface 192.168.200.20
LOC 23 45 32 N 89 23 18 W –24m 30m
Issues Only works if mapping between device and location is correct. Not secure/private
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197
Invisible Internet Telephony
VoIP technology will appear in . . . Internet appliances home security cameras, web cams 3G mobile terminals fire alarms chat/IM tools interactive multiplayer games
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198
VoIP Reliability & Manageability
Reliability: PSTN benchmarks… Work all the time, except for maintenance windows Faults: network, hardware, software Duplicated systems: no upgrade downtime Monitors, automatic failovers
Manageability: accurate and flexible billing systems, error reporting and resolution, call tracing, adds/moves/changes, Lack of network state (IP model) makes this
difficult => mediated calls (eg: softswitch etc reinstate some of this…)
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199
IPtel for appliances: “Presence”
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200
VoIP Standards (Enterprise View)
3rd PartyCall Servers &Gatekeepers
RTP
H.323 annex G,
SIP
H.323Gateway
SIPGateway Stimulus
Terminals
ThickTerminal
s
EnterpriseCall
Server
SIP H.248, Stimulus
H.323H.323
RTP
RTP
RTP
IP-enabledPBX/KS
H.248,Stimulus
SIP, H.323
H.323, SIP, Q.Sig
SIP, H.323
RTP
RTP
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VoIP Standards (Carrier View)
3rd PartyCall Agents &Gatekeepers
RTP
H.323, SIP-TBICC
Application/Media Server
SIPGateway
MegacoGateway
Softswitch/ Call Agent/
MGC
SIP Megaco/ H.248 MGCP
RTP
RTP
RTP
Signalling(SS7)
Gateway
SIP
Sigtrans, Q.BICC
MGCPGateway
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VoIP Summary: Big Picture