Selecting Standards That Will Be Implemented

Size: px

Start display at page:

Download "Selecting Standards That Will Be Implemented"

Barrie Wheeler
6 years ago
Views:

1 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 IP Telephony January 19, 2000

2 Carrier-Class IP Telephony 2 Overview Architectures: peer vs. master/slave Master/slave: MGCP and Megaco Peer-to-peer H.323 SIP services beyond signaling the standards process security issues

3 Carrier-Class IP Telephony 3 Architecture MG MGC Internet circuit-switched voice (POTS, ISDN) proxy GK GK proxy gateway PSTN RTP circuit-switched voice SIP H.323 Megaco/MGCP/MDCP

4 Carrier-Class IP Telephony 4 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

5 Carrier-Class IP Telephony 5 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

6 Carrier-Class IP Telephony 6 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 (finished) or Megaco/H.248 (evolving) 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

7 Carrier-Class IP Telephony 7 MGCP Architecture STP call agent SIP H.323 call agent SIP H.323 SS7 gwy TCAP ISUP SCP MGCP MGCP 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

8 Carrier-Class IP Telephony 8 IETF Protocol Stack MGCP/Megaco signaling quality of service media transport H.323 SDP SIP RTSP reservation RSVP measurement RTCP media encaps. (H.261, MPEG) RTP application daemon transport TCP UDP link network PPP IPv4, IPv6 physical AAL3/4 AAL5 PPP kernel Sonet ATM Ethernet V.34

9 Carrier-Class IP Telephony 9 IETF Architecture Call setup, registration: SIP, MGCP/Megaco Data transport: RTP, with RTCP for monitoring Voice mail: RTSP QOS resource reservation: RSVP, YESSIR,... Locating gateways: TRIP

10 Carrier-Class IP Telephony 10 SIP 101 SIP = signaling protocol for establishing sessions/calls/conferences/... session=audio,video,game,chat,... describedbysdp carried in SIP message 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

11 Carrier-Class IP Telephony 11 SIP Operation in Proxy Mode cs.columbia.edu cs.tu-berlin.de 1 INVITE 200 OK 7 2 henning? hgs@play location server 3 4 INVITE hgs@play 200 OK 6 5 play 8 ACK henning@columbia.edu tune 9 ACK hgs@play 10 media stream

12 Carrier-Class IP Telephony 12 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

13 Carrier-Class IP Telephony 13 SIP Advanced Features operation over UDP or TCP multicast invitations basic ACD interactive web response (IWR) UA $ proxy = proxy/redirect $ proxy/redirect stateless proxies: self-routing responses forking proxies: call several in sequence and/or parallel security: basic (password), digest (challenge/response), PGP

14 Carrier-Class IP Telephony 14 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

15 Carrier-Class IP Telephony 15 Internet Telephony as Part of Internet address = SIP address SIP URLs in web pages forward to , web page, chat session,... include web page in invitation response ( web IVR ) RTSP: choose your own music-on-hold include vcard, photo URL in invitation

16 Carrier-Class IP Telephony 16 standard since early 1999 SIP Status and Issues three bake-offs in 1999 for interoperability testing, about 30 companies attending work in progress: interaction with RSVP caller preferences ( no mobile phones, please ) call control interoperation with ISUP issue: SDP expressiveness

17 Carrier-Class IP Telephony 17 SIP Bake-Off Participants 3Com dynamicsoft Mitel 8x8 Ellemtel Netspeak Agilent Ericsson Nortel Alcatel Facet Nuera Broadsoft Helsinki Univ. OZ.com British Telecom Hewlett-Packard Pingtel Catapult Indigo Radcom Cisco IPcell Telogy Columbia University Lucent Vovida Dialogic MCI Worldcom VTEL Mediatrix

18 Carrier-Class IP Telephony 18 H.323 Components H.225.0: call control, RAS!GK: may I?, user location; RTP/RTCP H.235: security for H.323 terminals H.245: capabilities exchange, indications, notifications H.246: interoperability with PSTN H.332: large group conferences H.450: supplementary services H.246: interworking between H.323 and other H.xxx standards Q.931: call setup = ISDN, similar to Q.2931 (ATM) Q.932: supplementary services

19 Carrier-Class IP Telephony 19 H.323 derived from H.320 (ISDN multimedia) mostly ASN.1 (PER) based, but also Q.931 several versions, with support for all needed signaling TCP-based, except for H.323v3

20 Carrier-Class IP Telephony 20 H.323 Zones zone T1 GK GW T2 T3 R R Σ MCU

21 Carrier-Class IP Telephony 21 controls sessions H.323 gatekeeper performs user location and registration admission control reroutes signaling processes RAS (registration, admission, status) from H.323 terminals

22 Carrier-Class IP Telephony 22 Initialization: register with GK H.323 Phases GK admission: obtain permission; GK resolves address Call signaling: signaling connection to peer call initiation and completion/rejection Negotiation/configuration: negotiate roles during call capability exchange; determine mode of operation Media exchange: configure and open logical channels transmit and receive data streams Re-negotiation: change members, parameters, media,... Shutdown: terminate the call/conference deregister user on log-off

23 Carrier-Class IP Telephony 23 H.323v3 call setup caller RAS gatekeeper callee SETUP port 1720, 1300 CALL PROCEEDING ALERTING CONNECT audio 2 dynamic ports

24 Carrier-Class IP Telephony 24 H.323 problems already at version 4, most support version 2 very complex (200+ pages; 65 pages for call forwarding!) no multicast signaling limited multicast conferences ( MCUs) call = TCP connection $ mobility, reliability but: better capability negotiation (H.245) no media servers agile ports firewalls difficult

25 Carrier-Class IP Telephony 25 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, v3 to be decided.

26 Carrier-Class IP Telephony 26 H.323 vs. SIP Carrier Issues most hardware currently speaks H.323 longer term: bilingual, can coexist within same system interworking specification in progress, demonstrated SIP phone call NetMeeting

27 Carrier-Class IP Telephony 27 Primary Standards Organizations IETF: open process, Internet-focused: sip iptel avt enum megaco mmusic SIPextensions(callcontrol,callerpreferences,...) CPL, TRIP RTP E.164! IP DNS MEGACO/H.248 SAP, SDP ITU SG.16: H.323 and related work

28 Carrier-Class IP Telephony 28 Secondary (Standards) Organizations Softswitch Consortium: MGCP, SIP, RTP, RTSP; nominally, H.323 IMTC: inow!, ahit! interoperability clean up ETSI Tiphon: mostly European, glue documents, interoperability, architecture,...

29 Carrier-Class IP Telephony 29 Security Threats Theft of service: primarily applicable for resource reservation protocols Denial of service: repeated calls, bogus registrations to fill up database, bogus termination request,... Disclosure of sensitive data: call path, user identity Impersonation of identity: calls, registration

30 Carrier-Class IP Telephony 30 Security Protections lower-layer: SSL/TLS, IPsec but needs shared secret or certificate TLS! significant call setup delay unlike web server, every household needs certificate! some protection through randomness of call identifier (SIP), request sequence number or use of TCP connections SIP: password, digest (challenge/response) with shared secret, PGP for certificate-based security H.323: own security or TLS/IPsec (H.235)

31 Carrier-Class IP Telephony 31 Conclusion peer-to-peer always needed, master-slave for local use proliferation of standards bodies, but two do real work security issues require more attention

32 Carrier-Class IP Telephony 32 For more information... SIP: RTP: hgs/rtp Papers:

SIP: Status and Directions

SIP 1 SIP: Status and Directions Henning Schulzrinne Dept. of Computer Science Columbia University New York, New York schulzrinne@cs.columbia.edu Sylantro December 17, 1999 SIP 2 SIP overview/review SIP