1. 1 TAC2000/2000.7 802.16 IP Telephony Lab IP Telephony (Voice over IP) Assistant Professor Quincy Wu solomon@voip.edu.tw Graduate Institute of Communication Engineering National Chi Nan University

2. 2 TAC2000/2000.7 802.16 IP Telephony Lab Outline  Introduction to VoIP and SIP  Role of SIP in 3GPP IMS  Deployment of SIP Testbeds  Possible Issues

3. 3 TAC2000/2000.7 802.16 IP Telephony Lab Voice over UDP, not TCP  Speech Small packets, 10 – 40 ms Small packets, 10 – 40 ms Occasional packet loss is not a catastrophe Occasional packet loss is not a catastrophe Delay-sensitive Delay-sensitive  TCP: connection set-up, ack, retransmit → delays 5 % packet loss is acceptable if evenly spaced 5 % packet loss is acceptable if evenly spaced  Resource management and reservation techniques  A managed IP network In-sequence delivery In-sequence delivery  Mostly yes  But, UDP was not designed for voice traffic

4. 4 TAC2000/2000.7 802.16 IP Telephony Lab Real-Time Transport Protocol  Disadvantage of UDP Packets may be lost or out-of-sequence Packets may be lost or out-of-sequence  RTP: A Transport Protocol for Real-Time Applications RFC 1889; RFC 3550 RFC 1889; RFC 3550 RTP – Real-Time Transport Protocol RTP – Real-Time Transport Protocol RTCP – RTP Control Protocol RTCP – RTP Control Protocol  RTP over UDP A sequence number A sequence number A time stamp for synchronized play-out A time stamp for synchronized play-out Does not solve the problems; simply provides additional information Does not solve the problems; simply provides additional information

5. 5 TAC2000/2000.7 802.16 IP Telephony Lab RTP Header Format

6. 6 TAC2000/2000.7 802.16 IP Telephony Lab The RTP Header  Sequence number A random number generated by the sender at the beginning of a session A random number generated by the sender at the beginning of a session Incremented by one for each RTP packet Incremented by one for each RTP packet  Timestamp The receiver The receiver  Synchronized play-out  Calculate the jitter  Support silence suppression  The initial timestamp is a random number chosen by the sending application.  Payload Type (PT) In general, a single RTP packet will contain media coded according to only one payload format. In general, a single RTP packet will contain media coded according to only one payload format. RED is an exception. RED is an exception.

7. 7 TAC2000/2000.7 802.16 IP Telephony Lab RTP Payload Formats [1/2]  RTP carries the actual digitally encoded voice RTP header + a payload of voice/video samples RTP header + a payload of voice/video samples UDP and IP headers are attached UDP and IP headers are attached  Many voice- and video-coding standards A payload type identifier in the RTP header A payload type identifier in the RTP header  Specified in RFC 1890 and RFC 3551.  New coding schemes have become available A sender has no idea what coding schemes a receiver could handle. A sender has no idea what coding schemes a receiver could handle.  Negotiated by signaling protocols like SIP.

8. 8 TAC2000/2000.7 802.16 IP Telephony Lab Speech-coding Techniques  In general, coding techniques are such that speech quality degrades as bandwidth reduces. The relationship is not linear. The relationship is not linear. CodecBandwidthMOS CodecBandwidthMOS  G.71164kbps4.3  G.72632kbps4.0  G.723 6.3kbps3.8  G.72816kbps3.9  G.7298kbps4.0  GSM13kbps3.7  iLBC13.3kbps3.9

9. 9 TAC2000/2000.7 802.16 IP Telephony Lab SIP (Session Initiation Protocol)  A powerful alternative to H.323  More flexible, simpler  Easier to implement Advanced features Advanced features  Better suited to the support of intelligent user devices  A part of IETF multimedia data and control architecture

10. 10 TAC2000/2000.7 802.16 IP Telephony Lab SIP and Other Protocols  Originally Developed in the MMUSIC (Multiparty Multimedia Session Control) A separate SIP working group A separate SIP working group RFC 2543 RFC 2543 Many developers Many developers The latest version: RFC 3261 The latest version: RFC 3261  SIP + MGCP/MEGACO The VoIP signaling in the future The VoIP signaling in the future  SDP, RTSP (Real-Time Streaming Protocol), SAP (Session Announcement Protocol)

11. 11 TAC2000/2000.7 802.16 IP Telephony Lab Examples of SIP Messages

12. 12 TAC2000/2000.7 802.16 IP Telephony Lab SIP Servers [1/3] Registrar Registrar  Accepts SIP REGISTER requests –Indicating that the user is at a particular address –Personal mobility  Typically combined with a proxy or redirect server SIP Registrar Quincy@nctu.edu.tw 1.REGISTER registrar.nctu.edu.tw 2.Response

13. 13 TAC2000/2000.7 802.16 IP Telephony Lab SIP Servers [2/3] Proxy servers Proxy servers  Handle requests or forward requests to other servers  Can be used for call forwarding, time-of-day routing, or follow-me services

14. 14 TAC2000/2000.7 802.16 IP Telephony Lab SIP Servers [3/3] Redirect servers Redirect servers  Map the destination address to zero or more new addresses

15. 15 TAC2000/2000.7 802.16 IP Telephony Lab SIP Call Establishment [1/2]

16. 16 TAC2000/2000.7 802.16 IP Telephony Lab SIP History H.323SIP ITU-T protocol IETF protocol May 1995 Became “proposed standard” in March 1999. Study Group 16 SIP, SIPPING, SIMPLE WG Now V.5 Now RFC 3261

17. 17 TAC2000/2000.7 802.16 IP Telephony Lab ComponentsH.323SIP End Station Terminal SIP UA Network Server Gatekeeper Registrar, Redirect Server, Proxy Server MCU Conference Server PSTN Gateway

18. 18 TAC2000/2000.7 802.16 IP Telephony Lab Protocols H.323SIP SignalingRAS/Q.931SIP Capacity Negotiation H.245SDP CodecsAnyAny Real-time Communication RTP/RTCPRTP/RTCP

19. 19 TAC2000/2000.7 802.16 IP Telephony Lab Protocols (cont.) H.323SIP Message Encoding BinaryASCII Transport UDP and TCP Mostly TCP UDP and TCP Most UDP Data Conference T.120 Instant Message RFC 3428 Inter-Domain Routing Annex G DNS

20. 20 TAC2000/2000.7 802.16 IP Telephony Lab SIP is PBX/Centrex ready call waiting/multiple calls RFC 3261 hold RFC 3264 transfer RFC 3515/Replaces conference RFC 3261/callee caps message waiting message summary package call forward RFC 3261 call park RFC 3515/Replaces call pickup Replaces do not disturb RFC 3261 call blast RFC 3261 from Rohan Mahy’s VON Fall 2003 talk simultaneous ringing RFC 3261 basic shared lines dialog/reg. package barge-inJoin “ Take ” Replaces Shared-line “ privacy ” dialog package divert to admin RFC 3261 intercom URI convention auto attendant RFC 3261/2833 attendant console dialog package night service RFC 3261 centrex-style features boss/admin features attendant features

21. 21 TAC2000/2000.7 802.16 IP Telephony Lab Affordable SIP products  Finally, basic IP phones below $100  802.11 Wi-Fi phones  Video phones  Speakerphones

22. 22 TAC2000/2000.7 802.16 IP Telephony Lab 3GPP Networks  UMTS R4/R5 is an All-IP architecture with support for circuit- switchg terminals  Architecture based on GPRS with multimedia enhancements  MEGACO/H.248 is used for gateway control  SIP is used for establishing and terminating IP telephony calls  Support for integration of intelligent services

23. 23 TAC2000/2000.7 802.16 IP Telephony Lab

24. 24 TAC2000/2000.7 802.16 IP Telephony Lab NCNU VoIP Testbed

25. 25 TAC2000/2000.7 802.16 IP Telephony Lab Ad Hoc Network  Cellular Phone Home VoIP System 1 2 3 4 5 7 6 8

26. 26 TAC2000/2000.7 802.16 IP Telephony Lab Real-Time RTP Auditing  Libpcap  oRTP library  Windows Audio I/O

27. 27 TAC2000/2000.7 802.16 IP Telephony Lab Real-Time Audio Encryption  Real-time encryption algorithms  Analysis of various codecs  Schemes for key exchange  Secure RTP (RFC 3711)  Comparison with IPSec Encrypted RTP Media Stream Auditing ???

28. 28 TAC2000/2000.7 802.16 IP Telephony Lab Possible Issues  NAT Private IP addresses impose difficulty to end-to-end communications, although it works fine with client-server applications. Private IP addresses impose difficulty to end-to-end communications, although it works fine with client-server applications. Solutions: Solutions:  STUN (RFC 3489),  RTP Relay (a.k.a. Session Border Controller),  ICE (Interactive Connectivity Establishment, draft-ietf-mmusic-ice-19)  Firewall Some institute has strong security limitation. All UDP traffic (except DNS query) is prohibited. Some institute has strong security limitation. All UDP traffic (except DNS query) is prohibited. Solutions: Solutions:  Ask your IT staffs to add a rule for you  Skype’s success suggests that you may work out some way to traverse the firewall without the approval of IT staffs, but you are advised to make sure that doing so will not make them upset.

29. 29 TAC2000/2000.7 802.16 IP Telephony Lab Possible Issues (cont.)  SIP Servers Some labs may build their own SIP servers Some labs may build their own SIP servers For those who do not want to maintain an independent SIP server, they can register at other’s SIP server. For those who do not want to maintain an independent SIP server, they can register at other’s SIP server.  After the call setup, RTP audio streams go end-to-end directly.  SIP Phones Free softphones are available Free softphones are available  Windows Messenger  X-lite  Some labs also developed their own SIP softphone. Hardphone Hardphone  Do we want to buy some hardphone using WOAA budgets?  Is WiFi SIP Phone good for demonstration?