Introduction to VoIP. RFCs (RTP, SIP, H.323) Various books on VoIP

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1 Introduction to VoIP RFCs (RTP, SIP, H.323) Various books on VoIP S. Venkatesan Department of Computer Science Sprin 2008

2 Telephony Analo line Analo telephone network Manual operators on switchboard or Electromechanical ear Problems: Maintenance nihtmare Noise multiplex, diff voice circuits on diff freq S. Venkatesan Department of Computer Science Sprin 2008

3 Diital Telephony Analo line Central Office Diital lines Diital telephone network Diitize voice [8 bits/sample, 8000 samples/s Multiplex voice samples 125 microseconds 8 bits of call i S. Venkatesan Department of Computer Science Sprin 2008

4 Statistical MUX/Asynchronous Transmission Approximately 35% of time we talk; other 65% of time we listen and pause Why transmit silence? Transmit someone else s speech or data Need less bandwidth or pack more calls Uncertainty, jitter S. Venkatesan Department of Computer Science Sprin 2008

5 Call Quality 20% Traffic loss 10% Potentially Useful Poor 5% Good Toll Quality 100 ms 150 ms 400 ms Delay (one direction) S. Venkatesan Department of Computer Science Sprin 2008

6 Need for VoIP Voice is diitized and transmitted in current telecom networks anyway Interate voice and data (no need to build and maintain two separate networks) Make use of statistical multiplexin and use bandwidth more effectively Silence, mostly half-duplex conversation, etc, can be exploited. S. Venkatesan Department of Computer Science Sprin 2008

7 Main issues Encodin/decodin voice quality Loss of voice packets Variable delay of the IP network (between packets of the same conversation); jitter End-to-end delay (because of router delays, encodin/decodin delays), etc. Data and sinalin parts VoIP <> PSTN (Public Switched Telephone Network) S. Venkatesan Department of Computer Science Sprin 2008

8 Protocols Used (PC-PC call) PC IP Network PC Voice Voice RTP UDP IP Voice Packets SIP/H.323 TCP/UDP IP Sinalin packets S. Venkatesan Department of Computer Science Sprin 2008

9 Mic A/D Encoder Architecture Speaker D/A Decoder Jitter Buffer IP Stack IP Network S. Venkatesan Department of Computer Science Sprin 2008

10 RTP (Real-Time Transport Protocol) Receiver can compensate for jitter (for both voice and video) RTCP (Real Time Control Protocol) is used in conjunction with RTP (convey quality of transmission, such as averae packet loss, amount of jitter, etc.) RTP can be multicast (destination address = multicast address) S. Venkatesan Department of Computer Science Sprin 2008

11 RTP Defines a way to format IP packets carryin time-sensitive data Important fields: Type of data carried time-stamp of sampled data (to replay correctly with correct timin between coded packets) Sequence numbers (detect loss; copy last frame and repeat for aps or interpolate for missin frames) S. Venkatesan Department of Computer Science Sprin 2008

12 Payload Type Application determines contents Carry payload type in RTP headers Application does not need to examine contents. RTP Session: Association of participants (need two transport addresses, RTP/RTCP) S. Venkatesan Department of Computer Science Sprin 2008

13 RTP Header V (2) P(1) X(1) CC(4) M(1) PT(7) Sequence number (16) Time Stamp Synchronization Source Identifier Contributin Source Identifier 1 Contributin Source Identifier n Profile Dependent Size Data S. Venkatesan Department of Computer Science Sprin 2008

14 RTP Header Version (2 bits); currently = 2 P: Paddin bit: 1=payload has been padded for alinment (Last octet ives the number) X: extension bit. If 1, extension is present, fixed sinle xtn must follow header cc: Contributin SouRCe count M: Mark bit: If 1, this is the first packet after silence. (no packet durin silence) PayloadType: can vary dynamically 0=PCM mu law 8=PCM A law 9=G S. Venkatesan Department of Computer Science Sprin 2008

15 RTP Header Cont d Seq #: start at random number; increment by 1 for each packet time stamp[32 bits]: time value when fist sample in payload was sampled; initial value chosen randomly SSRC: ID of source of RTP stream CSRC: Contributin SouRCe (If RTP stream is a combination of several streams, ids of senders) S. Venkatesan Department of Computer Science Sprin 2008

16 RTCP Real-Time Transport Control Protocol Gathers stats about quality of network; periodically sends to sender of RTP packets Use: Adjust QoS parameters Types of reports: Sender Report (SR) (by active senders) Receiver Report (RR) (by participants that are not active senders) Source description-various parameters about SRC Bye - End of participation Application specific S. Venkatesan Department of Computer Science Sprin 2008

17 RTCP Header V (2) P(1) count (5) Type (8) Lenth (16) Data V: Version (Same version as RTP) P: Paddin; 1 = Data is padded. (Last byte shows # of padded bytes) Count: Number of reception blocks Type: Type of report Lenth: Lenth of this RTCP packet in 32 bit words minus one, includin the header and any paddin S. Venkatesan Department of Computer Science Sprin 2008

18 RTCP Header Version P (1=paddin used) RC (count): How many receiver blocks in packet PT: Payload type (200=SR, 201=RR, ) Lenth: Of sender report NTP Time Stamp: Network Time Protocol time stamp RTP Time Stamp Sender s packet count Sender s octet count S. Venkatesan Department of Computer Science Sprin 2008

19 RTCP Header Cont d RR Data: Fraction Lost Total number of packets lost Hihest sequence number received Interarrival jitter [averae] see next Last SR time stamp NTP time stamp received as part of most recent RTP sender report from SSRC Delay since last SR: number of (1/65536) seconds delay between receivin last SR from this SSRC and sendin this report S. Venkatesan Department of Computer Science Sprin 2008

20 Jitter Calculation Si = Time Stamp of sender of packet i Sj = Time Stamp of sender of packet j Ri = Time Stamp of receiver of packet i Rj = Time Stamp of receiver of packet j D[i,j]= (Rj-Ri)-(Sj-Si): Interarrival jitter Cumulative Jitter J = J+( D[i-1,j]) -J)/16 S. Venkatesan Department of Computer Science Sprin 2008

21 Echo Handlin Traditionally, 4 wire to 2 wire transformation adds echo Two types of echo: Electric echo Acoustic echo S. Venkatesan Department of Computer Science Sprin 2008

22 Acoustic Echo Time between speech and the same sound ettin into the mic: <=20 milliseconds? No noticeable echo >=40 ms? Problem Speaker phone is also a source of echo S. Venkatesan Department of Computer Science Sprin 2008

23 Echo Suppressers: Introduce lare loss in path from me to far end if far end person is talkin Squelch if both talk at the same time S. Venkatesan Department of Computer Science Sprin 2008

24 Echo Cancellers Build an estimate of echo (based on what oes out of the mic and the delay) Remove from incomin sinal at the correct time S. Venkatesan Department of Computer Science Sprin 2008

25 Sinalin Two sinalin protocols H.323 SIP [Session Initiation Protocol] S. Venkatesan Department of Computer Science Sprin 2008

26 Sinalin [H.323] End point: Entity makin/receivin calls Gate Keeper: Performs Address Translation, call control, etc. (one per LAN) End point (in our case PC) reisters with GateKeeper S. Venkatesan Department of Computer Science Sprin 2008

27 PC GRQ GCF/GRJ RRQ RCF/RRJ Gate Keeper Gate Keeper Request (ID) Gate Keeper Confirmation or Rejection with address (MAC+TSAP) of Keeper Reister Reistration Confirmation or Reject Reistration S. Venkatesan Department of Computer Science Sprin 2008

28 Call Sinalin PC Gate Keeper PC ARQ ACF/ARJ TCP connection for sinalin messaes Setup Call Processin (optional) ARQ Alertin Connect ACF/ARJ RTP Packets Alert User User answers S. Venkatesan Department of Computer Science Sprin 2008

29 DTMF: Dual Tone Multi-Frequency OK with G.711 (coder with no assumption about sinal bein sound) Other codecs: Cannot transmit DTMF s (Here accuracy is more important than timin) Special messaes carry IVR (interactive Voice Response) responses Special RTP loical channel to carry DTMF S. Venkatesan Department of Computer Science Sprin 2008

30 SIP (Session Initiation Protocol) Sinalin protocol to Initiate Manae and Terminate voice/video sessions on packet networks >=1 participants, unicast/multicast Text coded and can be extended S. Venkatesan Department of Computer Science Sprin 2008

31 SIP entities User aent Initiates/terminates sessions [client initiates SIP request, server serves SIP request] Proxy Server Both as client and server, interprets requests, can serve locally, translate/rewrite and forward Redirect Server Accept SIP request, map to >=0 new addresses and return to client Reistrar [like HLR] accepts REGISTER request, update database S. Venkatesan Department of Computer Science Sprin 2008

32 SIP Messaes Request and response SIP messaes are sent over UDP/TCP SIP is used with SDP (Session Description Protocol); SDP describes capabilities of the sender [vocoder used, etc.] S. Venkatesan Department of Computer Science Sprin 2008

33 SIP Messaes INVITE initiate session; Session description included in messae body ACK confirm session establishment BYE terminate connection CANCEL cancel pendin INVITE OPTIONS capability inquiry REGISTER bind permanent address to current location S. Venkatesan Department of Computer Science Sprin 2008

34 User Aent (Client) Invite Tryin Rinin OK User Aent (Server) Rin user Examples User picks up phone Disconnect Call Ack Bye OK Call connected S. Venkatesan Department of Computer Science Sprin 2008

35 PC <-> PSTN? PC IP PSTN IP-PSTN Gateway (Voice GW and Sinalin GW) S. Venkatesan Department of Computer Science Sprin 2008

36 PSTN <-> PSTN over IP? PSTN IP Network PSTN IP-PSTN Gateway IP-PSTN Gateway Voice packets Sinalin packets Call Aents S. Venkatesan Department of Computer Science Sprin 2008

37 IP-PSTN Gateway Services SCTP Sinalin GW Call Aent SS7 Call processin RTP MGW Voice trunks Media Gateway/Residential Gateway/ Access Gateway [many boxes] S. Venkatesan Department of Computer Science Sprin 2008

38 Bi Picture Accountin/ Authorization MGCP CA Subscriber data AGW RTP MGCP TGW Trunkin GW Voice Trunks+SS7 S. Venkatesan Department of Computer Science Sprin 2008

39 MGCP [Media Gateway Control Protocol] 8 MGCP commands: Notification request [CA->GW] Notify [tell CA when an event has occurred] CRCX [create connection] Modify Connection [chane parameters of a previously created connection] DLCX [Delete existin connection] Audit endpoint/connection Restart in proress [GW->CA; GW is oin down or in process of comin up] S. Venkatesan Department of Computer Science Sprin 2008

40 User Off hook Dial tone Res/Access Gateway Notification Request Ack Notify Ack MGCP Call Aent Initialization Routin DB Diits Notification Request Ack Notify Ack Notification request Ack Collect diits Send diits Stop collectin diits; Monitor for transitions S. Venkatesan Department of Computer Science Sprin 2008

41 User Res/Access Gateway CRCX Ack Call Aent Routin DB Trunkin GW Create connection; seize incomin circuit, Caller id, options Query Response CRCX Ack (with id of trunk ) MGCP Create a connection for New call with parameters Vocoder details, pack. Period, etc. S. Venkatesan Department of Computer Science Sprin 2008

42 User Res/Access Gateway Modify connection Call Aent Trunkin GW LEC IAM (has IP address of TGW, port#, RTP profile..] ACK Notification Request Ack ACM IAM ACM Rin the phone [caller can hear the rinin; What kind of rinin?] Initial Address Messae Address Completion Messae Remove Rinin Notification Request Ack ANM Stop rinin ANM Answer Messae; Called party answers MDCX Ack Connection was receive only; full duplex mode now S. Venkatesan Department of Computer Science Sprin 2008

43 User Res/Access Gateway Call Aent Trunkin GW LEC DLCX REL DLCX REL User disconnects call Ack Ack Off Hook Notify Ack Notification Request Ack Relay to Call Aent Reset endpoint S. Venkatesan Department of Computer Science Sprin 2008

Transporting Voice by Using IP

Transporting Voice by Using IP Voice over UDP, not TCP Speech Small packets, 10 40 ms Occasional packet loss is not a catastrophe Delay-sensitive TCP: connection set-up, ack, retransmit delays 5 % packet