Internet Telephony: VoIP, SIP & more

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 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

What is VoIP? Why VoIP? Where is VoIP Today?

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

What is VoIP? Protocol Soup
SDP MGCP SGCP SIP Megaco H.323 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 year’s model.” [Tanenbaum] Q.SIG Sigtrans H.GCP VPIM H.245

Telephony over IP standards bodies
ITU - International Telecommunication Union IETF - Internet Engineering Task Force. ETSI - European Telecommunications Standards Institute ANSI - American National Standards Institute TIA - Telecommunications Industry Association IEEE - Institute for Electrical and Electronics Engineers

Why VoIP? Telephony: Mature Industry
AT&T Divestiture

Why VoIP: Price/call plummeting due to overcapacity
AT&T Divestiture 1996 deregulation

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…

Why VoIP ? Data vs Voice Traffic
Note: quantity  quality  value-added Interactive 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?

Trends: Total Phone vs Data Revenues

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

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 Most minutes are still based on rate arbitrage vs. PSTN Future of VoIP services development is really going to based on IP originated services and advanced applications. Calls originating as IP gain cost advantages by traveling as far possible as on IP networks before PSTN interconnection. Best is IP termination. What are these new killer apps that are going to be? Combinations of Premises and Network based IP services like Virtual PBX or Centrex or presence based messaging that create virtual integrated communications . Services that integrate voice seamlessly with other mission critical IP apps like call centers. Integration of local, LD and International voice services on Global VPNs.

VoIP Volumes Are Accelerating While Adoption of Applications is Growing
M of Minutes VoIP VPN Traffic Enterprise Adoption of VoIP / IPT Applications Respondents North America Rest of the World M of Minutes Virtual PBX + Managed IP PBX traffic Source: Giga Group, "Next Generation IP Telephony Applications Deliver Strategic Business Value", October 20, 2003 Slide Title: VoIP Volumes Are Accelerating While Adoption of Applications are Growing Key message: Predicted growth for VoIP volumes is accelerating for both VPN and IP PBX. IPT applications will see dramatic growth as adoption increases. PROBE: Probe predicts that VoIP minutes over VPN will exceed 180 billion minutes worldwide by The VoIP minutes over Virtual PBX and Managed IP PBX combined will exceed 300 billion minutes. The total Voice-over-managed-service minutes will exceed 480 billion by 2007. Over half of the traffic will be outside North America from year 2005. In the same Probe report, it is also predicted that the market for VoIP VPN services in Asia will exceed that in the U.S. by 2007. Acronyms: IPT (IP Telephony); IPCC (IP Call Center ); Un. Comm (Unified Communications.); Un. Conf. (Unified Conferencing); 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 North America Rest of the World Source: Probe Research Inc.: Reaching the Big Guys + Global Enterprise Forecast. September 2002

Drivers Are Evolving From Cost Savings to Added Business Value…
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 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 2007 2006 2005 2004 2008 2002 Business Case Justification Based on Cost Savings V1 Apps V1 Apps - e.g. IP-PBX, Basic Call Functions, Branch offices, Toll-bypass Slide Title: Clients Drivers are Evolving From Cost Savings to Added Business Value Key Message: Research Indicates a Growing Market: Initially, migration to VoIP solutions were driven by cost justifications. As we move into 2004, businesses will begin to consider VoIP to future proof their technology investments. This will then evolve into leveraging the converged infrastructure to support applications that drive business value, such as unified communications and contact center functionality Note: This is a good opportunity to probe customer on what they are experiencing to better understand their situation COST SAVINGS: Toll By-Pass: Effective Use of Bandwidth: Personnel / Staffing efficiencies: Convergence / Consolidation: Decreased capital: 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 ADDITIONAL ANALYST QUOTES THAT REINFORCE KEY MESSAGE: 1) Business are requiring end-to-end VoIP solutions and management support to accelerate the implementation and benefits of VoIP or IP Telephony (The Yankee Group, September 2003) 2) The compound average growth rate (CAGR) for IP telephone revenue will be just under 30% through the period 2003 – 2007 (IDC, March 2003) 3) In-Stat/MDR predicts that 2003 will be the first year ever where there are more IP stations (telephone sets) sold than traditional PBX stations. (In-Stat/MDR, April-2003) The adoption of IP telephony grew by about 10 percent in Business are requiring end-to-end VoIP solutions and management support to accelerate the implementation and benefits of VoIP or IP Telephony (The Yankee Group, September 2003)

Summary: Why VoIP? Cost reduction: Operational Improvement:
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, and fax mail integration Mobility enabled by IP networking Web + Overseas Call Centers Collaborative applications New Integrated Applications 3Cs: “Convergence” & “Costs” & “Competition”

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

Consumer VoIP over broadband
Broadband Infrastructure Residential Media Gateway Media Gateway Controller Traditional phone Signaling and media gateways To reach PSTN or other networks

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

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 LocateMe, Do-Not-Disturb, Unified Portal

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

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

Enterprise: Private Branch Exchange (PBX)
Post-divestiture phenomenon... 7040 External line 7041 Corporate/Campus Telephone switch Private Branch Exchange Another switch 7042 7043 Corporate/Campus LAN Internet

Enterprise VoIP: Yesterday’s networks
Circuit Switched Networks (Voice) CO PBX PBX CO CO Headquarters Branch Offices Router Router “Yesterday’s networks” means what was being installed and designed in the past. Of course this also means they are the predominant design in today’s. Router Router Router Packet Switched Networks (IP)

Enterprise VoIP: Today’s networks Toll by-pass
Circuit Switched Networks (Voice) PBX CO PBX CO Headquarters CO Branch Offices Router “Today’s networks” means what one can safely/easily build today. Router Router Router Router Packet Switched Networks (IP)

Enterprise VoIP: Tomorrow’s networks Unified/Converged Networks
Legacy PSTN Router Router Router Router Router Unified Networks (Voice over IP) Headquarters Branch Offices

AT&T’s Integrated Infrastructure Supports Multiple Endpoints, Access Technologies and Application Services Voice Applications: IP Centrex, IP Call Center and Distant Worker 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 Call Control Element Common VoIP Connectivity Layer NG Border Elements SIP Border Elements H.323 Border Elements MGCP Border Elements IP/MPLS Converged Network SLIDE TITLE: AT&T’s Integrated Infrastructure Supports Multiple Endpoints, Access Technologies and Application Services The VoIP Connectivity Layer is built as a common and shared layer on top of the converged IP/MPLS network. The core network is surrounded by a multi-service access / multi-service edge network that supports all popular access technologies including TDM, ATM, Frame Relay and Ethernet. The VoIP infrastructure can be reached via any of these access technologies. For each type of access protocol, a corresponding Border Element manages the specifics of the protocol, provides a uniform view and flexibility to support multiple VoIP access protocols such as H.323, MGCP, H.248, SIP, as well as any VoIP protocol that may emerge in the future. This is achieved by surrounding the VoIP connectivity layer with Border Elements (BEs). BEs mark the trust boundaries and translate the specifics of various VoIP access protocols into Session Initiation Protocol (SIP) – the single common internal protocol used by all VoIP infrastructure components. AT&T’s Call Control Element (CCE) controls and manages the VoIP infrastructure and provides a single interface to application servers residing the Applications Layer. Working with various Bes, the CCE creates the removes call legs and joins call legs to establish connectivity between endpoints. SIP uses the Internet model for scalability - fast and simple in the core, smarter with less volume in the periphery. Allows for multi-media applications. AT&T is implementing SIP across the global networking infrastructure to lead the interoperability for multiple client based solutions. Security – A BE provides all the necessary security and screening for the customer sites it interacts with. It authenticates subscribers, customers and partners, and provides NAT and firewall functions as appropriate. MPLS offers numerous benefits: advanced application support with Class of Service (CoS) and Quality of Service (QoS) mechanisms, network scalability through any-to-any connectivity and communications flexibility through its support of secure IP VPNs AT&T was the first major networking provider to deliver MPLS services with the initial customer implementation in early Today, most major carriers have either deployed or announced plans to deploy MPLS. Security…MPLS Security – it is designed to provide a highly secure networking environment, while minimizing the risks associated with many potential threats DEFINITIONS: MGCP (Media Gateway Control Protocol)a protocol between a device and a service network (dumb device to smart network protocol). It is used for controlling telephony gateways from external call control elements called media gateway controllers or call agents. SIP (Session Initiation Protocol) – is the signaling protocol that the IP network uses to establish the call over the IP infrastructure. It does not dictate architecture. It’s an application layer control simple signaling protocol for VoIP implementations H.323: The H.323 standard provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS). PSTN SIP endpoints H.323 endpoints MGCP endpoints

VoIP Network Utilities Ensure 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 App. Server Media Server App. Server The purpose of this chart is to provide several basic call flow examples that require a VoIP Network Utility to ensure seamless business operations for our enterprise customers. These examples illustrate the interactions required in both the IP and Circuit Switched Networks that must reliably and predictably interwork to provide the end to end performance customers have come to expect from the circuit switched network. The architecture provides the modularity to efficiently scale to meet e-business demands while providing the foundation to support quick, efficient service introductions. Security and flow controls are implemented in both the IP and Circuit Switched Networks to ensure the customer’s applications are protected. Outbound Call (IP to Circuit Switched) Call Flow: IP Phone dials the Number IP Edge receives the call and provides access control (e.g., authentication, Firewall/NAT control, QOS control) and sends the call to the softswitch Softswitch determines the appropriate Application Server (AS) with the help of the service broker function and forwards the call to the AS (e.g., 8YY-AS) for handling the call. To simplify the chart the service broker is not shown. AS requests the Softswitch to route the call to the PSTN user Softswitch contacts the network routing engine (NRE) to find a Gateway to route the call to the PSTN. The NRE is not show in order to simplify the chart. Softswitch then forwards the call to the destination Gateway The destination Gateway completes the call over the PSTN knowing the phone number Inbound Call (Circuit Switched to IP) Call Flow: Circuit Switched phone dials the telephone number Circuit Switched network connects the call to the Gateway associated with that PSTN Number, or a range of PSTN Numbers Gateway receives the call and and sends the call to the Softswitch Softswitch determines the appropriate Application Server (AS) with the help of the service broker function AS requests the Softswitch to route the call to the IP user Softswitch contacts the network routing engine (NRE) to find a BE to route the call to the IP user Softswitch then forwards the call to the destination BE The destination BE completes the call over the IP access network knowing the phone number 800 Call (Circuit Switched to IP) Call Flow: Circuit Switched phone dials 800 telephone number NCP communicates with AS that supports Circuit Switched to IP call control interworking to determine endpoint for call. NCP directs call to appropriate BE/GW with appropriate endpoint information Softswitch determines the appropriate Application Server (AS) with the help of the service broker function and forwards the call to the AS that support 800 calls for handling the call Softswitch contacts the network routing engine (NRE) to find a BE to route the call to the IP user. NRE not shown in order to simplify chart. The destination Gateway completes the call over the IP access network associated with the termination number SDN Call (IP to Circuit Switched) Call Flow: IP Phone dials the SDN Number IP Edge receives the call and provides access control (e.g., authentication, Firewall/NAT control, QOS control) and sends the call to the Softswitch Softswitch determines the appropriate Application Server (AS) with the help of the service broker function and forwards the call to the AS that supports call control interworking for handling the call AS communicates with the NCP to determine the appropriate call treatment Softswitch contacts the network routing engine (NRE) to find a Gateway to route the call to the PSTN user Gateway Softswitch Redirect Call Circuit Switched to IP IP Network SDN Call IP to Circuit Switched

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 Two ways to unify: enable voice switches (PBX) to connect to IP, or enable IP devices (routers, L3 switches) to connect to voice. This slide shows the former by putting an IP interface (Ethernet) in the PBX. Switch

Telephony-enabled packet networks
Enterprise Router with telco interfaces T1/PRI BRI Branch office router with telco interfaces Analog trunk/line Analog “dongle” a few analog lines for fax/phone Central Office Router VoIP Gateway Two ways to unify: enable voice switches (PBX) to connect to IP, or enable IP devices (routers, L3 switches) to connect to voice. This slide shows the latter.

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

Voice over Packet: Market Forecast – North America

Telephony: History, Review & Trends

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]

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)

PSTN Evolution Office Switched Full Mesh Office Switched W/ Hierarchy
Started out with 2 phones connected by a copper wire. Each new phone meant another wire to every other telephone. Obviously, we had a scaling problem with this sort of architecture. Alexander Graham Bell solved this problem by creating a switching office. In this model, all the copper lines go to a central location where a live operator made physical patches to connect the users. As the number of switching offices grew it became desirable to connect them so that users could make long distance calls, so switching offices were interconnected. This eventually led to the same scaling problems as the full mesh topology had. Office Switched W/ Hierarchy Full Mesh Office Switched

AT&T Telephony Hierarchy
Class 1 Class 2 Eventually, the hierarchy evolved into five levels of switching offices, an architecture that remains essentially intact today. The traffic for each voice conversation is transmitted “up” to the level immediately above the current one in the hierarchy. Depending on its ultimate destination, the traffic continues upstream toward the long-distance network or starts back down a different lower level path to the recipient. Each switching office is named by its level in the hierarchy, with the offices closest to your home called Class 5 offices, the next closest Class 4 offices, etc. Today only the Class 5 and Class 4 names remain in common use, although Class 3, Class 2, and Class 1 switching offices do exist. With the breakup of AT&T in 1984, the network was opened up for other carriers to start providing voice services. Companies like Sprint and MCI started building their own toll networks and companies like mine started deploying end office switches. Class 3 Class 4 Class 5 Source: Computer Networks, Andrew S. Tanenbaum

PSTN early days 40s-60s In-band signaling: voice and control channel same Complex and dedicated hardware Hard to add new apps like caller-id, 800 calling etc Tandem Office Local Office Local Office User A User B

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

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 Central Office (CO) Analog Digital Analog

The PSTN – Digitization
Voice frequency is Hz, with the main portion from 300 – 3400 Hz Nyquist Theorem states that sampling must be done at twice the highest frequency to recreate Hz was chosen as the maximum frequency, thus sampling at 8000 Hz PCM = 8kHz * 8 bits per sample = 64 kbit/s

Quantization

Companding

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

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) = Mbit/sec = T-1 Channel 1 Channel 2 Framing Bit Channel 3 TDM Channel 1 Channel 2 Channel 3 … Channel 24 … Channel 24 1 D4 Frame

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

Legacy Digital Circuit Switch
Centralized Intelligence Proprietary Code Proprietary service deployment Very expensive

What’s the difference between a Class 5 and a Class 4 switch?
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 Basic architecture is the same. Services and placement are different.

The PSTN – NANP (212) 555 4210 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 IXC 212 LEC 555 4210 PSTN (212)

The PSTN – Call Routing Both NANP and International Numbering Plan – E.164, use prefix-based dialing SS7 212 5644 408 555 PSTN 5644 The ‘555’ LEC switch then checks the station code and signals the appropriate phone The first LEC receives a call, seeing ‘1’ as the first digit and then passing the call on to the IXC switch. The IXC then routes the call to the remote IXC responsible for ‘212’ The ‘212’ IXC looks at the office code and passes it on to the ‘555’ LEC switch

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)

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 independents interconnected & expanded

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

US Telephone Network Structure (after 1984)
Eg: AT&T, Sprint, MCI Eg: SBC, Verizon, BellSouth

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

VoIP Technologies

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

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 / G IP Gateway Voice gets transmitted via RTP over an IP infrastructure

Internet Multimedia Protocol Stack

Packet Encapsulation RTP datagram UDP datagram IP packet
Version, flags & CC Payload Type Sequence Number Timestamp Synchronization Source ID CSRC ID (if any) Codec Data UDP datagram Source Port Number Destination Port Number UDP length UDP checksum Data Version & header length Protocol IP packet TOS Total Length Packet ID Flags & Frag Offset TTL Header Checksum Source Address Destination Address Options (if any) Data Ethernet overhead = = 26 bytes IP overhead = = 20 bytes UDP overhead = = 8 bytes RTP overhead = = 12 bytes Ethernet Inter-frame gap is not considered. (another 12 bytes) Start of frame delimiter Length or Ethertype Ethernet Frame Inter-frame gap Preamble Destination Address Source Address Data Pad Checksum

RTP – Real-time Transport Protocol
RTP datagram Version, flags & CC Payload Type Sequence Number Timestamp Synchronization Source ID CSRC ID (if any) Codec Data 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

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)

Signaling: VoIP Camps Our focus H.323 SIP “Softswitch” BICC
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”

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

SIP LAN Interface RTP IP Audio Codec G.711 G.723 G.729 Video Codec
H.261 H.263 RTP RTCP SIP TCP UDP IP LAN Interface

SIP functionality IETF-standardized peer-to-peer signaling protocol (RFC 2543): Locate user given -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

SIP Addresses Food Chain

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

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 4 5

SIP: Personal Mobility
Users maintain a single externally visible identifier regardless of their network location

Expand existing PBXs w/ IP phones
Transparently …

SIP as Event Notification Protocol

SIP: Presence

Light-weight signaling: Session Initiation Protocol (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)

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)

SIP-based Architecture
rtspd SIP/RTSP Unified messaging RTSP media server sipum Quicktime RTSP clients RTSP T1/E1 RTP/SIP Telephone Cisco 2600 gateway switch SIP conference server sipconf Web based configuration Web server SIP proxy, redirect server SQL database sipd e*phone sipc Software SIP user agents Hardware Internet (SIP) phones SIPH.323 convertor NetMeeting sip323 H.323

Example Call Bob signs up for the service from the web as sipd canonicalizes the destination to He registers from multiple phones sipd rings both e*phone and sipc Bob accepts the call from sipc and starts talking Alice tries to reach Bob INVITE Web based configuration Web server Call Bob SIP proxy, redirect server SQL database sipd e*phone sipc Software SIP user agents Hardware Internet (SIP) phones ecse.rpi.edu

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

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.

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.

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).

SIP Design Choices

Proxy Server us.gov parliament.uk
1. INVITE SIP/2.0 From: 2. INVITE SIP/2.0 From: 3. SIP/ ok From: us.gov parliament.uk Location Server 1 & 5 4 2 & 6 3 george.w.bush 5. ACK SIP/2.0 From: 4. SIP/ OK From: 6. ACK SIP/2.0 From: Proxy server

Redirect Server us.gov parliament.uk 1 & 3 2 5 4 & 6
Location Server george.w.bush parliament.uk 1 & 3 2 5 4 & 6 3. ACK From: 1. INVITE From: 2. SIP/ Moved temporarily Contact: 5. SIP/ OK To: 4. INVITE From: 6. ACK From: Redirect Server

Assumes Endpoints(Clients) know each other’s IP addresses
SIP Call Signaling Assumes Endpoints(Clients) know each other’s IP addresses SIP Endpoint SIP Gateway Signaling Plane Invite SIP + SDP (TCP or UDP) 180 Ringing 200 OK Ack RTP Stream Bearer Plane RTP Stream Media (UDP) RTCP Stream

PSTN to IP Call 1 2 3 5 4 PBX PSTN External T1/CAS Regular phone
(internal) Call 1 Gateway Internal T1/CAS (Ext: ) Call 7134 2 SIP server sipd Ethernet 3 sipc 5 Bob’s phone SQL database 4 7134 => bob

IP to PSTN Call 4 5 3 1 2 PBX Internal T1/CAS Call 85551212
Regular phone (internal, 7054) PSTN External T1/CAS Call 5 Gateway ( ) 3 Ethernet SIP server sipd sipc 1 Bob calls SQL database 2 Use

Traditional voice mail system
Dial Alice Bob Phone is ringing .. The person is not available now please leave a message ... ... Your voice message ... Disconnect Bob can listen to his voice mails by dialing some number.

SIP-based Voicemail Architecture
phone1.office.com Bob INVITE INVITE INVITE Alice sipd REGISTER vm.office.com The voice mail server registers with the SIP proxy, sipd Alice calls through SIP proxy. SIP proxy forks the request to Bob’s phone as well as to a voic server.

Voicemail Architecture
phone1.office.com; Bob CANCEL Alice sipd 200 OK 200 OK RTP/RTCP v-mail vm.office.com; After 10 seconds vm contacts the RTSP server for recording. SETUP vm accepts the call. Sipd cancels the other branch and ... rtspd ...accepts the call from Alice. Now user message gets recorded

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, … )

Registrar & Proxy or Redirect Server *Gateway PSTN, ISDN, ATM, etc System Management admission control address translation/forwarding Firewall bypassing Interface to non-IP or H.323 networks *User Agent *User Agent *User Agent *Endpoints Media streams: RTP/RTCP (G.911, G.723.1, … ) Conferencing does not need another box (MCU) End-user devices and network proxies

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

SDP: Session Description Protocol
Not really a protocol – describes data carried by other protocols Used by SAP, SIP, RTSP, H.332, PINT. Eg: v=0 o=g.bell IN IP s=Come here, Watson! u= c=IN IP b=CT:64 t= k=clear:manhole cover m=audio 3456 RTP/AVP 96 a=rtpmap:96 VDVI/8000/1 m=video 3458 RTP/AVP 31 m=application udp wb

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

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

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

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)

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.

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 Via: SIP/2.0/UDP server1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp SIGCOMP is the compression protocol

Device Configuration

SIP Scaling Issues

SIP Scaling (contd) SIP Load Characteristics:

SIP vs H.323 vs Megaco

Terminal Control/Devices Terminal Control/Devices
H vs SIP Typical UserAgent Protocol stack for Internet Terminal Control/Devices Terminal Control/Devices Q.931 H.245 RAS RTCP RTCP Codecs SIP SDP Codecs RTP RTP TPKT TCP UDP Transport Layer IP and lower layers Typical user agent stack for H.323 and SIP look like these. The call signaling and user registration part are taken care by Q.931/RAS and Sip respectively. H.323 uses H.245 for media description and logical channel signaling, whereas SIP user agents, almost always, use SDP. RTP/RTCP is used by both to carry media traffic.

SIP versus H.323 H.323 and SIP are direct competitors in peer-level call control space H.323 SIP ITU-T SG-16 IETF SIP Stds Body 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) 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 Industry Acceptance

SIP-H.323: Interworking Problems Eg: Call setup translation
Q.931 SETUP INVITE Destination address 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 Problems in call setup translation: Three pieces of information needed for call setup : Destination signaling address Self and remote media capabilities Self and remote media transport address SIP carries these in INVITE and its response. H.323 spreads them across different stages. Mapping multistage dialing in H.323 to single stage in SIP is not trivial. H.323 v2 Fast-Start supports single stage dialing, but it is optional.. H.323: Multi-stage dialing

H.323 Standard Series LAN Interface RTP IP Audio Codec G.711 G.723
Video Codec H.261 H.263 System Control H.245 Control Data Interface T.120 H.225 Call Setup RTP RTCP RAS Gatekeeper TCP UDP This is a simplified version of the standard H.323 stack drawing that appears everywhere. The basic point of it is to show that there are multiple standards for some functions (such as audio codecs), but that all the audio and video data streams happen over RTP (Real-time transport Protocol) which runs on UDP (User Datagram Protocol) which means it’s unacknowledged and connectionless. Whereas the control protocols all operate on TCP which is an acknowledged, connection-oriented protocol. This makes sense because if a voice packet is lost there’s no real point in re-sending it (it would be too late), but you want to make sure a connect message or key press (DTMF) got through. IP LAN Interface

Internet Telephony Protocols: H.323

H.323 (contd) Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs)

H.323 Model - Gatekeeper Routed Call
RAS RAS Call Setup/Signaling Call Setup/Signaling Call Control Call Control So how does it work? There are 4 basic groups of functions: Registration/Admission/Status (RAS) for security/bandwidth monitoring/address translation, Call Setup to create the call, and Call control to determine capabilities and send control data, and the actual voice traffic. H.323 encompasses other things as well, including conferencing and video standards, but we’ll focus on the endpoint/gateway/gatekeeper functions in this presentation. Voice Channel Endpoint Gateway

H.323 Model - Gatekeeper Direct Call
RAS RAS Call Setup/Signaling So how does it work? There are 4 basic groups of functions: Registration/Admission/Status (RAS) for security/bandwidth monitoring/address translation, Call Setup to create the call, and Call control to determine capabilities and send control data, and the actual voice traffic. H.323 encompasses other things as well, including conferencing and video standards, but we’ll focus on the endpoint/gateway/gatekeeper functions in this presentation. Call Control Voice Channel Endpoint Gateway

H.323 Call Signaling Assumes Endpoints(Clients)
know each other’s 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 Open Logical Channel H.245 (TCP) This is actually a simplified view of H.323’s call procedure. In reality, the call setup stage takes an average of 6 packets (TCP connection creation, TCP ACK, setup, alerting, call proceeding, and connect) and possibly more. After that, the negotiation mechanism for codec channels, master-slave determination, etc. can take 8 or more as well. H.323 version 2 allows for a fast connect method whereby the voice channels are set-up during the call-setup stage (the info is actually in the setup messages) which makes it much faster. Unfortunately, some devices only do fast-start, and others only do slow-start and thereby cannot interoperate. Open Logical Channel & Acknowledge Open Logical Channel Acknowledge RTP Stream Bearer Plane RTP Stream Media (UDP) RTCP Stream H.323v1 (5/96) - 7 or 8 Round Trips H.323v2 Fast Start (2/98) - 2 Round Trips

ITU-T H.323 Architecture Tour
Gate Keeper (GK) *Gateway (GW) PSTN, ISDN, ATM, etc *Multipoint Control Unit (MCU) Multipoint Controller (MC) Processor (MP) *Terminal *Terminal *Terminal *Endpoints Media streams: RTP/RTCP (G.911, G.723.1, … )

Gate Keeper (GK) *Gateway (GW) PSTN, ISDN, ATM, etc *Multipoint Control Unit (MCU) Multipoint Controller (MC) Processor (MP) System Management zone management b/w management & admission control address translation centralized control (“gatekeeper control mode”) Interface to non-IP networks *Terminal *Terminal *Terminal *Endpoints Media streams: RTP/RTCP (G.911, G.723.1, … ) Conferencing End-user devices and network proxies

Gate Keeper (GK) Components of the H.323 protocol suite: Q.931 = ISDN call signalling H = 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 RAS H CS H.245 CC H.450.x SS *Gateway (GW) PSTN, ISDN, ATM, etc *Multipoint Control Unit (MCU) Multipoint Controller (MC) Processor (MP) *Terminal *Terminal *Terminal *Endpoints Media streams: RTP/RTCP (G.911, G.723.1, … )

Gatekeeper Routed Call
1. Setup called: caller: :: 2. Setup called: :: caller: 3. Connect Atlanta Zone (404) 2, 6, 10, 14 1, 5, 9, 13 Gatekeeper 4, 8, 12, 16 3, 7, 11, 15 4. Connect 9. Open Channel G.729/30ms, :6400 5. TCS media: G.711/30ms, G.729/30ms 10. Open Channel G.729/30ms, :6400 6. TCS media: G.711/30ms, G.729/30ms 11. Open Channel G.729/20ms, :2300 7. TCS media: G.729/20ms, G.723 12. Open Channel G.729/20ms, :2300 8. TCS media: G.729/20ms, G.723 13. ACK 14. ACK 15. ACK 16. ACK

Gatekeeper Direct Call
1. ARQ called: caller: :: 2. ACF called: :: 3. Setup called: caller: :: Atlanta Zone (404) 1 2 Gatekeeper 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, :6400 8. Open Channel G.729/20ms, :2300

MEGACO/H.248, Softswitch Concepts

Master/Slave vs. Peer Comparison
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 Lowest cost end device Higher cost end device Cost Lower performance “local” services Sometimes higher performance distributed services (e.g.. call control) Higher performance local services High performance User Interface Performance 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 Feature deployment Update servers only Services can come and go dynamically Update / download all end devices in network (yikes!) Features more static per-device Protocols MEGACO/H.248, MGCP H.323, SIP

Megaco/H.248 LAN Interface RTP IP Audio Codec G.711 G.723 G.729
Video Codec H.261 H.263 RTP RTCP Megaco TCP UDP IP LAN Interface

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 ITU SG-16 initiates gateway control project, H.GCP starting from MDCP (May 99) Megaco/H.248 Agreement reached between ITU SG16 and IETF Megaco to work together to create one standard (Summer 99)

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 Call Model Termination + Connection P2P Single Media Single Media Conferencing Terminations Physical & Ephemeral Command Grouping Ad hoc Embedding Event Quarantine Event Packages (MGCP) Media Session Description SDP Protocol Encoding Text Transport UDP Bold entries indicate additional features in Megaco vs. MGCP

Megaco Architecture Whirlwind Tour
Call Agent Media Gateway Controller Signalling Gateway PSTN, ATM, etc trunks lines SS7 etc Sigtran Analog Media Gateway PSTN trunking PSTN line IP Phone Megaco Scope Signalling Gateway Layer (SG) Interface to SS7 signalling etc Not in Megaco scope (IETF Sigtran) Endpoint (e.g.. H.323 Gateway, Terminal, MCU) Gateway Function Call control (e.g.. H.323, SIP…) 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

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) Media GW Controller Media Gateway Device control 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

Telephone/Residential
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 Virtual Switch Media Gateways SS7 Signalling Gateway Sigtrans PSTN Trunking Media Gateway Media GW Controller RTP H.323 Device RTP PSTN Line Media Gateway Megaco/ H.248 Telephone/Residential Media Gateway SIP Device Cable Modem Media Gateway

Megaco call in action (optional)
MG1 MG2 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 “ ” Reply: Notify Add: TDM to RTP, what codecs? Reply: Add, codec G.729

Megaco call in action (continued)
MG1 MG2 MGC Add: TDM to RTP, ring phone Phone Rings Reply: Add Modify: ip of MG2, ringback Hears Ring Reply: Modify Off-Hook 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 Disconnect Subtract: TDM and RTP Subtract:TDM and RTP Reply: Subtract Reply: Subtract

Megaco/H.248 IP Phone Control
Cisco’s Skinny, Nortel’s UNIStim, etc., are very similar protocols but they’re not interoperable H.323 GW MGC H.323 Voice (RTP) In theory the RTP stream should go direct phone<->GW, but many today tandem through the MGC Media, LCD, Softkey Control Media, LCD, Softkey Control Voice (RTP) Voice (RTP) Voice (RTP) So how does it work? There are 4 basic groups of functions: Registration/Admission/Status (RAS) for security/bandwidth monitoring/address translation, Call Setup to create the call, and Call control to determine capabilities and send control data, and the actual voice traffic. H.323 encompasses other things as well, including conferencing and video standards, but we’ll focus on the endpoint/gateway/gatekeeper functions in this presentation. Voice (RTP) IP Phone Media Gateway IP Phone Media Gateway

Vendor Support for Standards
Source: Network World and Mier Communications - August, 2001

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

MGCP/H.248/Megaco Media Gateway Controller (MGC) SIP
Master/Slave MGCP Media Gateway Signaling Gateway Media 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

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

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

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

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

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

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

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

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

MGC and MG Roles Media Gateway Controller Media Gateway
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

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

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

Softswitch – Application Server
Application Servers(AS) Can be broken apart and distributed in the network LDAP Directory Server Feature Server COPS Corba Network Elements Policy Server SIP Directory Server- Directory Services, Embedded Databases, Network databases, IP Address Registar Media Serve r- Announcements, Voice Prompts, Voice Recognition Feature Server - Class 5 Voice services, Unified Messaging Policy Server – Location,Translation, Routing, Quality of Service, Security Management Server- OSS Interface, Billing, Order Management and provisioning, Network Management, System Diagnostics, Subscriber Self Provisioning Connectivity Server - Session management, Connection management, Connectivity to rest of network Corba Media Server Management Server Connectivity Server SIP,Parlay,JAIN

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

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

SIP vs MEGACO: Summary

SIP vs MEGACO (contd)

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)

Speech Coding and Speech Coders for VoIP

Taxonomy of Speech 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 kbps Vocoders: 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)

Speech Quality of Various Coders

Speech Quality (Contd)

Actual Bandwidth Used

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)

Pulse Amplitude Modulation (PAM)

Pulse Code Modulation (PCM)
* PCM = PAM + quantization

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

How it works for T1 Lines Companding blocks are shared by all 16 channels

Recall: Taxonomy of Speech 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 kbps Vocoders: 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

Vocoders Encode only perceptually important aspects of speech w/ fewer bits than waveform coders: eg: power spectrum vs time-domain accuracy

LPC Analysis/Synthesis

Speech Generation in LPC

CELP Encoder

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).

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 is HORRIBLE (continued from previous slide) For G.723 encoding, packets are sent approx. every 30msec, with size being fixed at Ethernet frame size of 82 bytes = IP of 64 bytes = UDP of 44 = RTP of 24 bytes. But if the voice is silent, then no packets are sent.

Packet Encapsulation RTP datagram UDP datagram IP packet
Version, flags & CC Payload Type Sequence Number Timestamp Synchronization Source ID CSRC ID (if any) Codec Data UDP datagram Source Port Number Destination Port Number UDP length UDP checksum Data Version & header length Protocol IP packet TOS Total Length Packet ID Flags & Frag Offset TTL Header Checksum Source Address Destination Address Options (if any) Data Ethernet overhead = = 26 bytes IP overhead = = 20 bytes UDP overhead = = 8 bytes RTP overhead = = 12 bytes Ethernet Inter-frame gap is not considered. (another 12 bytes) Start of frame delimiter Length or Ethertype Ethernet Frame Inter-frame gap Preamble Destination Address Source Address Data Pad Checksum

G.711 (10ms) Clear Channel Voice
80 byte voice bundles RTP Frame 12 RTP Header Voice Payload 80 2 80 UDP Datagram Voice Payload Destination Source Length Checksum 12 RTP Header 4 12 80 IP Packet Header Source Voice Payload UDP Header RTP Header 8 Type Destination CRC. IP into Ethernet 8 6 6 2 120 4 Preamble Source IP Payload 1 2 120 2 1 IP into Frame Relay Flag IP Payload Flag Address Frame Check + IP into ATM 5 48 5 48 + 5 24 16 8 Header IP Payload Header IP Payload Header IP Payload Padding Trailer

30 byte voice bundles RTP Frame 12 RTP Header Voice Payload 30 2 30 UDP Datagram Voice Payload Destination Source Length Checksum 12 RTP Header 4 12 30 IP Packet Header Source Voice Payload UDP Header RTP Header 8 Type Destination CRC. IP into Ethernet 8 6 6 2 70 4 Preamble Source IP Payload 1 2 70 2 1 IP into Frame Relay Flag IP Payload Flag Address Frame Check + IP into ATM 5 48 5 22 18 8 Header IP Payload Header IP Payload Padding Trailer

20 byte voice bundles RTP Frame 12 RTP Header Voice Payload 20 2 20 UDP Datagram Voice Payload Destination Source Length Checksum 12 RTP Header 4 12 20 IP Packet Header Source Voice Payload UDP Header RTP Header 8 Type Destination CRC. IP into Ethernet 8 6 6 2 60 4 Preamble Source IP Payload 1 2 60 2 1 IP into Frame Relay Flag IP Payload Flag Address Frame Check + IP into ATM 5 48 5 12 28 8 Header IP Payload Header IP Payload Padding Trailer

G.723.1 (30ms) Clear Channel Voice
20-24 byte voice bundles 12 20-24 RTP Frame 2 20-24 UDP Datagram Voice Payload Destination Source Length Checksum 12 RTP Header RTP Header Voice Payload 4 12 20-24 IP Packet Header Source Voice Payload UDP Header RTP Header 8 Type Destination CRC. IP into Ethernet 8 6 6 2 60-64 4 Preamble Source IP Payload 1 2 60-64 2 1 IP into Frame Relay Flag IP Payload Flag Address Frame Check IP into ATM 5 48 + 5 12-16 28-24 8 Header IP Payload Header IP Payload Padding Trailer

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

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

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.

Voice codecs: Summary G.711 uncompressed PCM audio stream
8ks/s of 8 bit values = 64kbps packet “sizes” = 10, 20, 30 and 60ms G Wideband (7kHz) G.726 ADPCM - 10,20,30,60ms - 32kbps G.723.1 MLQ - 30ms or 6.3kbps Silence suppression G.729 CS-ACELP - 10, 20, 30ms - 8kbps Annex B adds silence suppression Within codecs, different packet sizes/sample sizes can be chosen (10ms, 20ms, 30ms, and 60ms are common ones). A packet “size” really refers to how long a sample is taken for, or how many fixed-size packets are stuffed into an IP frame. Generally, the larger the packet is, the greater the effects of delay and loss will be, but the less overhead it will take. This becomes important for software-based devices, which cannot handle the processing power needed to send out frames every 10ms - they typically use a 60ms frame size which means they send larger frames every 60ms. (sampling at N millisecond intervals would also mean sending a frame every N milliseconds, unless the codec uses an overlap or redundancy scheme) The rates shown above are not really the data rate seen on the wire, since it does not account for IP and Ethernet overhead. For G.711 encoding, packet sizes for 10ms: RTP length of 80 = UDP length of 100 = IP frame of 120 = Ethernet frame of 146 bytes. 20ms: RTP length of 160 = UDP length of 180bytes = IP frame of 200 = Ethernet frame size of 226 bytes. 30ms: RTP of 240 = UDP of 260 = IP of 280 = Ethernet frame of 306 bytes. 60ms: RTP of 480 bytes = UDP of 500 = IP of 520 = Ethernet frame of 546 bytes. Note: this includes the preamble and start frame delimiter of the Ethernet frame, but not the inter-packet gap. For G.729 it’s 10ms: RTP of 10 bytes = UDP of 30 bytes = IP of 50 = Ethernet of 76 bytes. So the UDP CBR = 24 kbps. 20ms: RTP of 20 bytes = UDP of 40 = IP of 60 = Ethernet of 86 bytes. 30ms: RTP of 30 bytes = UDP of 50 = IP of 70 = Ethernet of 96 bytes.

Recap: Speech Quality of Various Coders

Miscl: Other standards, ENUM, E-911, Presence etc

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 SS7 Network Signaling Gateway Signaling Gateway Sigtrans SS7 The Internet CO Sigtrans Sigtrans Virtual Switch Trunk Gateway D-channel Sigtrans Media Gateway Signalling Gateway SIP, H.323 PRI Megaco/ H.248 B-channels Media GW Controller PBX Virtual Switch RTP

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 - Presentation 5 - Session User Adaptation Modules 4 - Transport SCTP 3 - Network IP 2 - Link MLPPP / FR / ATM 1 - Physical Ethernet / SONET/Serial

(1) SS7 Signaling Using IP Transport
The IETF M2UA MTP2-User Adaptation Layer from the Sigtran WG SSP SSP Applications Applications STP TCAP ISUP TCAP ISUP SCCP SCCP MTP3 MTP3* MTP3 MTP2 MTP2 M2UA M2UA SCTP SCTP IP IP

(2) SS7 / IP Interworking The IETF M3UA MTP3-User Adaptation Layer
from the Sigtran WG SSP MGC SS7 SG Call Processing Application Call Processing Application Nodal Inter-working Function ISUP ISUP MTP3 MTP3 M3UA M3UA MTP2 MTP2 SCTP SCTP IP IP

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 BICC ISN BICC ISN BICC ISUP ISUP PSTN PSTN TDM TDM Class 4 Switch Class 4 Switch Data Network

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 voic systems Adds more useful info: vcard, signature, multiple addresses POP3 still used to download voic to your favorite client (Outlook, Eudora, Pine, etc.) POP3 Browser VPIM PBX SIP/H.323 Plain Phone Unified Messaging System Unified Messaging System SIP Device

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 The gateway location and routing problem has is considered one of the more difficult problems in IP telephony. The selection of an egress gateway for a telephony call, traversing an IP network towards an ultimate destination in the PSTN, is driven in large part by the policies of the various parties along the path, and by the relationships established between these parties. As such, a global directory of egress gateways in which users look up destination phone numbers is not a feasible solution. Rather, information about the availability of egress gateways is exchanged between providers, and subject to policy, made available locally and then propagated to other providers in other ITADs, thus creating routes towards these egress gateways. This would allow each provider to create its own database of reachable phone numbers and the associated routes - such a database could be very different for each provider depending on policy.

Midcom (Middlebox Communication)
1. INVITE SIP/2.0 From: 2. INVITE SIP/2.0 From: 3. SIP/ ok From: us.gov parliament.uk Location Server 1 & 5 4 2 & 6 3 george.w.bush 5. ACK SIP/2.0 From: 4. SIP/ OK From: 6. ACK SIP/2.0 From: Proxy server Firewall 3.5 Midcom Protocol

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)

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. 01:45 Normal disc.

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 tolerant Local cache Roll back

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.

ENUM vs DNS DNS (or internet) names: interpreted right to left:
Eg: Telephone numbers: interpreted left to right: Eg: ENUM: (RFC 3761) telephone numbers written DNS-style, Rooted at the domain e164.arpa. So, becomes 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 as

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, …

E911 - Requirements 911 Services
Power stays on when building power fails Need callers phone number and location Services must be modified during a 911 call Disable call-waiting Disable three-party calls Caller cannot hangup and place another call PSAP – Public Safety Answering Point

E911 – VoIP Enhancements VoIP has the potential of enhancing E911 functionality Multimedia communication Audio – emulate existing services Video – images and/or biometrics to/from emergency technicians Text – for hearing impaired Call setup could contain medical background Can be locally maintained, does not a master database Calls can easily be forwarded or transferred Fast call setup times PSAP could easily be deployed or relocated anywhere Internet access is available.

E911 – Using DNS to convey location
Based on network device name pigface GL S3.US “110 Stony Point Rd.,Santa Rosa CA” Based on Geographic location (longitude/latitude) GPOS Binary (includes precision indicator) LOC N W –24m 30m Issues Only works if mapping between device and location is correct. Not secure/private RFC 1876

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

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…)

IPtel for appliances: “Presence”

VoIP Standards (Enterprise View)
H.323 annex G, SIP Enterprise Call Server H.323, SIP, Q.Sig 3rd Party Call Servers & Gatekeepers IP-enabled PBX/KS H.248, Stimulus H.323 H.323 SIP H.248, Stimulus SIP, H.323 SIP, H.323 Thick Terminals SIP Gateway H.323 Gateway RTP RTP RTP This looks bad, but really as long as each standard encompasses unique functions (I.e., as long as each doesn’t overlap), it’s ok. Problem is, some overlap: H.323 and SIP, a bit of MGCP and H.323. Another thing to note is that this many protocols means it is a much bigger deal training the network support staff than “just another protocol on IP”. Stimulus Terminals RTP RTP RTP

VoIP Standards (Carrier View)
H.323, SIP-T BICC Softswitch/ Call Agent/ MGC Sigtrans, Q.BICC Signalling (SS7) Gateway 3rd Party Call Agents & Gatekeepers SIP SIP Megaco/ H.248 MGCP SIP Gateway RTP RTP Application/ Media Server RTP This looks bad, but really as long as each standard encompasses unique functions (i.e., as long as they don’t overlap), it’s ok. The problem is, some overlap: H.323 and SIP, a bit of MGCP and H.323. Another thing to note is that this many protocols means that it is a much bigger deal training the network support staff than “just another protocol on IP”. Megaco Gateway MGCP Gateway RTP

VoIP Summary: Big Picture

Internet Telephony: VoIP, SIP & more

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