1 TAC2000/2000.7 802.16 IP Telephony Lab IP Telephony (Voice over IP) Assistant Professor Quincy Wu...
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Transcript of 1 TAC2000/2000.7 802.16 IP Telephony Lab IP Telephony (Voice over IP) Assistant Professor Quincy Wu...
11 TAC2000/2000.7
802.16 IP Telephony Lab
IP Telephony (Voice over IP)
Assistant Professor Quincy [email protected]
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
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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 [email protected]
REGISTER sip:registrar.work.com SIP/ 2.0Via: SIP/ 2.0/ UDP station1.work.com; branch=z9hG4bK123Max-Forwards: 70From: sip:[email protected]; tag = 123456To: sip:[email protected]: [email protected]: 1 REGISTERContact: sip:[email protected]: 7200Content-Length: 0
SIP/ 2.0 200 OKVia: SIP/ 2.0/ UDP station1.work.com; branch=z9hG4bK123From: sip:[email protected]; tag = 123456To: sip:[email protected]: [email protected]: 1 REGISTERContact: sip:[email protected]: 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
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
[email protected] [email protected]
1.Request [email protected]
2.Request [email protected]
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 [email protected]
2.Moved temporarily Contact: [email protected]
3.ACK
4.Request
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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2424 TAC2000/2000.7
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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
2525 TAC2000/2000.7
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
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802.16 IP Telephony Lab
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?