The Effect of Standards on the Growth of IP Telephony

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1 hgs/me 1 The Effect of Standards on the Growth of IP Telephony Henning Schulzrinne Dept. of Computer Science Columbia University New York, New York schulzrinne@cs.columbia.edu IP Telephony Evolution Palm Springs, CA January 30, 2001

2 hgs/me 2 Overview Architectures: peer vs. master/slave Master/slave control: MGCP and Megaco Peer-to-peer signaling: H.323 SIP Mapping between numbers and services: ENUM and TRIP Event notification and presence Programming multimedia services

3 hgs/me 3 IETF VoIP Protocol Architecture Languages/APIs JAIN CPL voicexml Parlay servlets sip cgi Directory/Discovery DNS/enumLDAP TRIP SLP peer to peer Signaling SIP MGCP PINT H.248 SPIRITS master slave RTSP DiffServ QoS IntServ RTP Transport TLS SCTP

4 hgs/me 4 IETF VoIP Protocol Architecture: Goals Leverage content-neutrality of Internet more than just voice and legacy services video, shared applications, multi-party text chat Imperceptible transition between communication modalities Extensible to presence, instant messaging and event notification Centrex lesson: user-controlled services Allow services in end systems and network servers Multiple levels of security: IPSec, TLS, application-layer

5 hgs/me 5 Differences: Internet Telephony $ POTS separate control, transport (UDP) no triangle routing separate connectivity from resource availability separate services from bit transport datagram service less bootstrapping in-band signaling higher speed features ônetworkö! end system: distinctive ringing, caller id, speed dialing, number translation,... scaling features: intra-pbx = inter-lata and general protocols: user-network = network-network signaling

6 hgs/me 6 PSTN Legacies to Avoid E.164 numbers might as well wear bar codes tones and announcements in-band signaling for features (DTMF) systems with user-interface knowledge (12 keys, voice) voice-only orientation (BICC, MGCP/Megaco) integration of bit transport and services service-specific billing separate signaling & billing trusted networks without crypto confine PSTN knowledge to edge of network

7 hgs/me 7 Invisible Internet Telephony VoIP technology will appear in Internet appliances home security cameras, web cams 3G mobile terminals fire alarms and building sensors chat/im tools interactive multiplayer games 3D worlds: proximity triggers call

8 hgs/me 8 Carrier and Enterprise VoIP Traditionally, separate signaling: ISDN, CAS vs. ISUP service restrictions, e.g., CF inefficient Now, largely the same: hosted ( ASP ), run own servers or combinations carrier: multiple domains per server if not outsourced, TRIP for gateway selection

9 hgs/me 9 Peer-to-Peer Architecture IP telephones, gateways, PCs with software = IP hosts may use servers (H.323 gatekeepers, SIP proxy servers) end system fully state-aware protocols for call setup: H.323 or SIP more flexible user interface

10 hgs/me 10 Implementing Services end system server caller id x call forwarding, follow me x x three-way calling x distinctive ringing x 69 x? no solicitation x x do not disturb x x call curfew? x

11 hgs/me 11 Master-Slave Architecture master-slave: MGC controls one or more gateways allows splitting of signaling and media functionality please send audio from circuit 42 to uses MGCP (implemented) or Megaco/H.248 (standardized, but just beginning to be implemented) gateway can be residential basis of PacketCable NCS (network control system) architecture service creation similar to digital PBX or switch! can charge for caller id, call waiting

12 hgs/me 12 MGCP Architecture STP call agent MG controller SIP call agent SIP TCAP MG controller SS7 gwy H.323 H.323 ISUP SCP MGCP/Megaco MGCP/Megaco SS7 RGW RGW Internet RTP TGW PSTN for all but small system, need peer-to-peer! MGCP system can call SIP or H.323 end system all use RTP to transfer data

13 hgs/me 13 SIP 101 SIP = signaling protocol for establishing sessions/calls/conferences/... session=audio,video,game,chat,... describedbysdpcarriedinsipmessage 1. called server may map name to user@host 2. callee accepts, rejects, (! forward new address) 3. if new address, go to step 1 4. if accept, caller confirms 5....conversation caller or callee sends BYE

14 hgs/me 14 SIP Operation in Proxy Mode cs.tu-berlin.de 1 INVITE 200 OK 7 2 henning? cs.columbia.edu location server hgs@play 3 4 INVITE hgs@play 200 OK 6 5 play 8 ACK henning@columbia.edu tune 9 ACK hgs@play 10 media stream

15 hgs/me 15 SIP Operation in Redirect Mode ieee.org? location server tu-berlin.de 4 1 INVITE henning@ieee.org 302 Moved temporarily Contact: hgs@columbia.edu 2 henning columbia.edu ACK henning@ieee.org INVITE hgs@columbia.edu columbia.edu OK ACK hgs@columbia.edu hgs

hgs/me 16 SIP Personal Mobility alice@columbia.edu (also used by bob@columbia.edu) yahoo.com alice17@yahoo.

16 hgs/me 16 SIP Personal Mobility (also used by yahoo.com tel: columbia.edu tel:

17 hgs/me 17 More SIP Internet Telephony Services camp-on without holding a line short message service ( instant messaging ) schedule call into the future call with expiration date add/remove parties to/from call mesh buddy lists

18 hgs/me 18 Internet Telephony as Part of Internet universal identifier: address = SIP address = IM address SIP URLs in web pages forward to , web page, chat session,... include web page in invitation response ( web IVR ) third-party control of calls via scripts, include vcard, photo URL in invitation user-programmable services: CGI (RFC 3050), CPL, servlets

19 hgs/me 19 SIP Status and Issues standard since early 1999, editorial revision in progress six bake-offs since 1999 for interoperability testing, about 60 companies attending work in progress: interaction with resource reservation caller preferences ( no mobile phones, please ) interoperation with ISUP ( SIP-T ) call transfer and third-party control conferencing: central server, end system, full mesh server benchmarking and scaling

20 hgs/me 20 SIP H.323 Comparison H.323 SIP Architecture stack element Conference control yes no Protocol mostly TCP mostly UDP Encoding ASN.1, Q.931 HTTPish Emphasis telephony multimedia, multicast Address flat alias, E.164 SIP, E.164 URLs Both SIP and H.323 are evolving: SIP additions, H.323v2 implements some SIP features.

21 hgs/me 21 A Signaling Protocol Prediction H.323 continues to be used for simple gateways, conferencing systems most large vendors produce both IP PBXs (IP phones) primarily SIP SIP as core signaling protocol for 3G networks less use of MGCP/Megaco, except for very large gateways BICC, SCTP only for legacy-burdened carriers

22 hgs/me 22 Legacy Signaling Transport Signaling Control Transport Protocol (SCTP) like TCP, but with enhancements for out-of-order delivery and multiple end points generally useful (e.g., can carry SIP) primarily useful for ISUP transport over IP does not mean use of ISUP for setting up VoIP sessions SIP in the middle : carry ISUP in message body

23 hgs/me 23 ENUM: Mapping E.164 Numbers to URLs Map single E.164 number to one or more URLs (e.g., SIP, mailto, H.323) transition from E.164 numbers to SIP URLs voice mail drop address number portability implementation enum database DNS e164.arpa SCP enum 00 1 IP INVITE sip:alice@wonderland.com

24 hgs/me 24 Mapping E.164 Numbers to URLs ENUM not suited for finding gateways: would need entry for every number single mapping only no ability to convey parameters Problems to be solved: Who manages country-level domains? Blocks of numbers? Who owns a number - customer or carrier? Slamming of services

25 hgs/me 25 Determining the Best VoIP Gateway Every gateway can reach almost any E.164 address pick closest, cheapest, preferred signaling, ITAD 1 ITAD 2 GW 00 1 IP LS EU IP 01 0 TRIP 01 0 IP... IP 01 0 ITAD 3

hgs/me 26 Example: Columbia CS Phone System Expand

connectivity Cisco 7960 MySQL user database LDAP

media server Sun Solaris PC Linux/FreeBSD/NT RTSP

sipd proxy/redirect server sipum PBX T1/E1 RTP SIP

26 hgs/me 26 Example: Columbia CS Phone System Expand existing PBX via IP phones, with transparent connectivity Cisco 7960 MySQL user database LDAP server sipconf conferencing server (MCU) rtspd RTSP media server Sun Solaris PC Linux/FreeBSD/NT RTSP unified messaging server Nortel Meridian Cisco 2600 sipd proxy/redirect server sipum PBX T1/E1 RTP SIP PhoneJack interface e*phone sipc plug n sip SIP H.323 converter sip h b wireless

27 hgs/me 27 The Largest Signaling Network Does Not Run SS7 AT&T: 280 million calls a day AOL: 110 million s/day, total about 18 billion/day total > 1 billion instant messages/day (AOL: 500 million) telephony signaling ß IM, presence

28 hgs/me 28 Commonalities Between Signaling and Events presence is just a special case of an event: Alice logged in ß house temperature dropped below 50 deg. need to locate mobile end points (for notifications) may need to find several different destinations same addressing for users presence often precursor to calls may replace call back and call waiting likely to be found in same devices events already in VoIP: message alert, call events, conf. joins

29 hgs/me 29 SIP as a Presence & Event Platform minimal SIP extension: SUBSCRIBE to request notifcations, NOTIFY when event occurs also, MESSAGE for IM, sessions for multi-party chats transition to true chat (and video) services such as reaching mobile phone while in meeting

30 hgs/me 30 Events: SIP for Appliances SUBSCRIBE SIP user agent NOTIFY DO INVITE SIP proxy (RGW)

31 hgs/me 31 Programming Internet Multimedia Services Primarily, creation, forwarding, proxying, rejection of calls APIs (Parlay, JAIN): protocol-neutral (SIP, H.323, ISUP), but may be least common denominator SIP CGI: use Perl and other scripting languages; easy to learn Servlets: Java only; faster than cgi; limited functionality CPL: = XML-based language for user service creation; portable across providers, but not all services Protocol-neutral: Parlay, JAIN, CPL Call creation: Parlay, JAIN VoiceXML is for voice-service creation after call setup

32 hgs/me 32 Conclusion basic IETF-based architecture in place SIP as foundation for services extensions for mobility, emergency services,...inprogress first (and last?) chance to recover from 120 years of legacy avoid replication of PSTN on packets most VoIP applications won t look like telephones range of engagement and asynchronicity, from call to IM to challenge of mobile services

33 hgs/me 33 For more information... SIP: RTP: hgs/rtp Papers:

Selecting Standards That Will Be Implemented

Carrier-Class IP Telephony 1 Selecting Standards That Will Be Implemented Henning Schulzrinne Dept. of Computer Science Columbia University New York, New York schulzrinne@cs.columbia.edu ICM Carrier Class