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802.16 IP Telephony Lab

IP Telephony (Voice over IP)

Assistant Professor Quincy Wusolomon@voip.edu.tw

Graduate Institute of Communication EngineeringNational Chi Nan University

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Outline

Introduction to VoIP and SIPIntroduction to VoIP and SIPRole of SIP in 3GPP IMSRole of SIP in 3GPP IMSDeployment of SIP TestbedsDeployment of SIP TestbedsPossible IssuesPossible Issues

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Voice over UDP, not TCP SpeechSpeech

Small packets, 10 – 40 msSmall packets, 10 – 40 ms Occasional packet loss is not a catastropheOccasional packet loss is not a catastrophe Delay-sensitiveDelay-sensitive

TCP: connection set-up, ack, retransmit → delaysTCP: connection set-up, ack, retransmit → delays

5 % packet loss is acceptable if evenly spaced5 % packet loss is acceptable if evenly spaced Resource management and reservation techniquesResource management and reservation techniques A managed IP networkA managed IP network

In-sequence deliveryIn-sequence delivery Mostly yesMostly yes

But, UDP was not designed for voice trafficBut, UDP was not designed for voice traffic

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Real-Time Transport Protocol

Disadvantage of UDPDisadvantage of UDP Packets may be lost or out-of-sequencePackets may be lost or out-of-sequence

RTP: A Transport Protocol for Real-Time ApplicationsRTP: A Transport Protocol for Real-Time Applications RFC 1889; RFC 3550RFC 1889; RFC 3550 RTP – Real-Time Transport ProtocolRTP – Real-Time Transport Protocol RTCP – RTP Control Protocol RTCP – RTP Control Protocol

RTP over UDPRTP over UDP A sequence numberA sequence number A time stamp for synchronized play-outA time stamp for synchronized play-out Does not solve the problems; simply provides additional informationDoes not solve the problems; simply provides additional information

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RTP Header Format

00

01

02

03

04

05

06

07

08

09

10

11

12

13

14

15

16

17

18

19

20

21

22

23

24

25

26

27

28

29

30

31

V=2 P X CC M PT Sequence Number

Timestamp

Synchronization Source (SSRC) Identifier

Countributing Source (CSRC) Identifier (0 to 15 entries)

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The RTP Header Sequence numberSequence number

A random number generated by the sender at the beginning of a sessionA random number generated by the sender at the beginning of a session Incremented by one for each RTP packetIncremented by one for each RTP packet

TimestampTimestamp The receiverThe receiver

Synchronized play-outSynchronized play-out Calculate the jitterCalculate the jitter Support silence suppressionSupport silence suppression The initial timestamp is a random number chosen by the sending application.The initial timestamp is a random number chosen by the sending application.

Payload Type (PT)Payload Type (PT) In general, a single RTP packet will contain media coded according to In general, a single RTP packet will contain media coded according to

only one payload format.only one payload format. RED is an exception.RED is an exception.

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RTP Payload Formats [1/2]

RTP carries the actual digitally encoded voiceRTP carries the actual digitally encoded voice RTP header + a payload of voice/video samplesRTP header + a payload of voice/video samples UDP and IP headers are attachedUDP and IP headers are attached

Many voice- and video-coding standardsMany voice- and video-coding standards A payload type identifier in the RTP headerA payload type identifier in the RTP header

Specified in RFC 1890 and RFC 3551.Specified in RFC 1890 and RFC 3551. New coding schemes have become availableNew 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.Negotiated by signaling protocols like SIP.

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Speech-coding Techniques

In general, coding techniques are such that speech In general, coding techniques are such that speech quality degrades as bandwidth reduces.quality degrades as bandwidth reduces.The relationship is not linear.The relationship is not linear.

CodecCodec BandwidthBandwidth MOSMOS G.711G.711 64kbps64kbps 4.34.3 G.726G.726 32kbps32kbps 4.04.0 G.723 G.723 6.3kbps6.3kbps 3.83.8 G.728G.728 16kbps16kbps 3.93.9 G.729G.729 8kbps8kbps 4.04.0 GSMGSM 13kbps13kbps 3.73.7 iLBCiLBC 13.3kbps13.3kbps 3.93.9

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SIP (Session Initiation Protocol) A powerful alternative to H.323A powerful alternative to H.323 More flexible, simplerMore flexible, simpler Easier to implementEasier to implement

Advanced featuresAdvanced features

Better suited to the support of intelligent user devicesBetter suited to the support of intelligent user devices A part of IETF multimedia data and control architectureA part of IETF multimedia data and control architecture

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SIP and Other Protocols Originally Developed in the MMUSIC (Multiparty Multimedia Originally Developed in the MMUSIC (Multiparty Multimedia

Session Control)Session Control) A separate SIP working groupA separate SIP working group RFC 2543RFC 2543 Many developersMany developers The latest version: RFC 3261The latest version: RFC 3261

SIP + MGCP/MEGACOSIP + MGCP/MEGACO The VoIP signaling in the futureThe VoIP signaling in the future

SDP, RTSP (Real-Time Streaming Protocol), SAP (Session SDP, RTSP (Real-Time Streaming Protocol), SAP (Session Announcement Protocol)Announcement Protocol)

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Examples of SIP MessagesRegistrarCollins@station1.work.com

REGISTER sip:registrar.work.com SIP/ 2.0Via: SIP/ 2.0/ UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: sip:Collins@work.com; tag = 123456To: sip:Collins@work.comCall-ID: 123456@station1.work.comCSeq: 1 REGISTERContact: sip:Collins@station1.work.comExpires: 7200Content-Length: 0

SIP/ 2.0 200 OKVia: SIP/ 2.0/ UDP station1.work.com; branch=z9hG4bK123From: sip:Collins@work.com; tag = 123456To: sip:Collins@work.comCall-ID: 123456@station1.work.comCSeq: 1 REGISTERContact: sip:Collins@station1.work.comExpires: 3600Content-Length: 0

a

b

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SIP Servers [1/3] RegistrarRegistrar

Accepts SIP REGISTER requestsAccepts SIP REGISTER requests– Indicating that the user is at a particular addressIndicating that the user is at a particular address

– Personal mobilityPersonal mobility

Typically combined with a proxy or redirect serverTypically combined with a proxy or redirect server

SIPRegistrarQuincy@nctu.edu.tw

1.REGISTERregistrar.nctu.edu.tw

2.Response

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SIP Servers [2/3]

Proxy serversProxy servers Handle requests or forward requests to other serversHandle requests or forward requests to other servers Can be used for call forwarding, time-of-day routing, or follow-me Can be used for call forwarding, time-of-day routing, or follow-me

servicesservices

SIPProxyCaller@work.com Quincy@home.net

1.Request Quincy@nctu.edu.tw

2.Request Quincy@home.net

4.Response 3.Response

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SIP Servers [3/3] Redirect serversRedirect servers

Map the destination address to zero or more new addressesMap the destination address to zero or more new addresses

Redirect Server

1.Request Quincy@nctu.edu.tw

Caller@work.com

Quincy@home.net

2.Moved temporarily Contact: Quincy@home.net

3.ACK

4.Request

Quincy@home.net

5.Response

6. ACK

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SIP Call Establishment [1/2]

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SIP History

H.323H.323 SIPSIP

ITU-T protocolITU-T protocol IETF protocolIETF protocol

May 1995May 1995 Became “proposed standard” in March Became “proposed standard” in March 1999.1999.

Study Group 16Study Group 16 SIP, SIPPING, SIMPLE WGSIP, SIPPING, SIMPLE WG

Now V.5Now V.5 Now RFC 3261Now RFC 3261

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Components

H.323H.323 SIPSIP

End StationEnd Station TerminalTerminal SIP UASIP UA

Network Network ServerServer

GatekeeperGatekeeper Registrar, Redirect Server, Registrar, Redirect Server, Proxy ServerProxy Server

MCUMCU Conference ServerConference Server

PSTN GatewayPSTN Gateway PSTN GatewayPSTN Gateway

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Protocols

H.323H.323 SIPSIP

SignalingSignaling RAS/Q.931RAS/Q.931 SIPSIP

Capacity NegotiationCapacity Negotiation H.245H.245 SDPSDP

CodecsCodecs AnyAny AnyAny

Real-time Real-time CommunicationCommunication

RTP/RTCPRTP/RTCP RTP/RTCPRTP/RTCP

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Protocols (cont.)H.323H.323 SIPSIP

Message EncodingMessage Encoding BinaryBinary ASCIIASCII

TransportTransport UDP and TCPUDP and TCP

Mostly TCPMostly TCP

UDP and TCPUDP and TCP

Most UDPMost UDP

Data ConferenceData Conference T.120T.120

Instant MessageInstant Message RFC 3428RFC 3428

Inter-Domain Inter-Domain RoutingRouting

Annex GAnnex G DNSDNS

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SIP is PBX/Centrex readycall waiting/multiple callscall waiting/multiple calls RFC 3261RFC 3261

holdhold RFC 3264RFC 3264

transfertransfer RFC 3515/ReplacesRFC 3515/Replaces

conferenceconference RFC 3261/callee capsRFC 3261/callee caps

message waitingmessage waiting message summary packagemessage summary package

call forwardcall forward RFC 3261RFC 3261

call parkcall park RFC 3515/ReplacesRFC 3515/Replaces

call pickupcall pickup ReplacesReplaces

do not disturbdo not disturb RFC 3261RFC 3261

call blastcall blast RFC 3261RFC 3261

from Rohan Mahy’s VON Fall 2003 talk

simultaneous ringingsimultaneous ringing RFC 3261RFC 3261

basic shared linesbasic shared lines dialog/reg. packagedialog/reg. package

barge-inbarge-in JoinJoin

““Take”Take” ReplacesReplaces

Shared-line “privacy”Shared-line “privacy” dialog packagedialog package

divert to admindivert to admin RFC 3261RFC 3261

intercom intercom URI conventionURI convention

auto attendantauto attendant RFC 3261/2833RFC 3261/2833

attendant consoleattendant console dialog packagedialog package

night servicenight service RFC 3261RFC 3261

centr

ex-s

tyle

featu

res

boss/admin features

attendant features

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Affordable SIP products

Finally, basic IP phones below Finally, basic IP phones below $100$100

802.11 Wi-Fi phones802.11 Wi-Fi phones Video phonesVideo phones SpeakerphonesSpeakerphones

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3GPP Networks UMTS R4/R5 is an All-IP architecture with support for circuit-UMTS R4/R5 is an All-IP architecture with support for circuit-

switchg terminalsswitchg terminals Architecture based on GPRS with multimedia enhancementsArchitecture based on GPRS with multimedia enhancements MEGACO/H.248 is used for gateway controlMEGACO/H.248 is used for gateway control SIP is used for establishing and terminating IP telephony callsSIP is used for establishing and terminating IP telephony calls Support for integration of intelligent servicesSupport for integration of intelligent services

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NCNU PBX

Phone31842

Phone31924

Phone59237

Phone59238

SIP Proxy Server

SIP Phone0944003005SIP Phone

0944003004

PSTN Gateway

SIP Phone0944002002

Phone4161

Phone4162

Phone4131

Phone4762

Phone049-2912360

StationInterface

StationInterface

StationInterface

StationInterface

Phone03-5912312

Admin Console

SIP Phone0944003003

SIP Phone0944002003

Hsinchu

Nantou

TrunkInterface

03-5712121

049-2910960

TrunkInterface

SIP Proxy Server

TANet

NCTU

PSTN

NCNUEdge Router

Edge Router

NCTU PBX

SoftphoneWLANAP

PSTN Gateway

NCNU VoIP Testbed

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802.16 IP Telephony LabAd Hoc Network Cellular Phone

Internet

802.11AccessPoint

PHS

SIP REGISTRAR STUN

Server

SIP PROXY

VoIP System

VoIP phone

WGSNGPRSBase

Station

PHSBase

Station

GSM

GSMBase

Station

PSTN

PHSHandset

MN1

MN2

MN3 MN4

Home VoIP System

1

2

3

4

5

7

6

8

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NCNU PBX

SIP Proxy Server

PSTN Gateway

SIP Phone0944002002

StationInterface

StationInterface

Phone31924

Phone31533

Phone31842

Phone31350

StationInterface

Real-Time RTP Auditing

SIP Phone0944002003

SIP Proxy Server

TANet

NCTU

NCNUEdge Router

Edge Router

NCTU PBX

Softphone

WiMAXBS

PSTN Gateway

Real-Time RTP Auditing

LibpcapLibpcap oRTP libraryoRTP library Windows Audio Windows Audio

I/OI/O

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Possible Issues NATNAT

Private IP addresses impose difficulty to end-to-end communications, Private IP addresses impose difficulty to end-to-end communications, although it works fine with client-server applications.although it works fine with client-server applications.

Solutions: Solutions: STUN (RFC 3489), STUN (RFC 3489), RTP Relay (a.k.a. Session Border Controller), RTP Relay (a.k.a. Session Border Controller), ICE (Interactive Connectivity Establishment, draft-ietf-mmusic-ice-19)ICE (Interactive Connectivity Establishment, draft-ietf-mmusic-ice-19)

FirewallFirewall Some institute has strong security limitation. All UDP traffic (except Some institute has strong security limitation. All UDP traffic (except

DNS query) is prohibited.DNS query) is prohibited. Solutions:Solutions:

Ask your IT staffs to add a rule for youAsk your IT staffs to add a rule for you Skype’s success suggests that you may work out some way to traverse the 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 firewall without the approval of IT staffs, but you are advised to make sure that doing so will not make them upset.that doing so will not make them upset.

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Possible Issues (cont.) SIP ServersSIP Servers

Some labs may build their own SIP serversSome labs may build their own SIP servers For those who do not want to maintain an independent SIP server, they For those who do not want to maintain an independent SIP server, they

can register at other’s SIP server.can register at other’s SIP server. After the call setup, RTP audio streams go end-to-end directly.After the call setup, RTP audio streams go end-to-end directly.

SIP PhonesSIP Phones Free softphones are availableFree softphones are available

Windows MessengerWindows Messenger X-liteX-lite Some labs also developed their own SIP softphone.Some labs also developed their own SIP softphone.

HardphoneHardphone Do we want to buy some hardphone using WOAA budgets?Do we want to buy some hardphone using WOAA budgets? Is WiFi SIP Phone good for demonstration?Is WiFi SIP Phone good for demonstration?