2. Copyrights © 2006. All rights Reserved. Introduction to VoIP Chetan Vaity August 2006

3. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Lets make some VoIP calls… Broadvoice Indian PSTN US phone Indian phone Internet 1 US PSTN 2 3

4. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com What is VoIP  Transfer of voice conversations over an IP based network  Also known as:  IP Telephony  Internet telephony  Broadband telephony  Voice over Broadband

5. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Essentials  What happens in a VoIP call?  Establish connection with the target  Various protocols  Capture voice, digitize and encode  Codecs  Transfer over network  Network issues  Interface with PSTN  Decode and reproduce voice

6. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Protocols  Signaling protocols  SIP ( I nternet E ngineering T ask F orce)  H.323 ( I nternational T elecommunications U nion)  All voice/video communications are done over separate transport protocols, typically RTP  Media protocols  RTP  RTCP

7. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Protocols – SIP  Session Initiation Protocol  SIP is primarily used in setting up and tearing down voice or video calls  SIP clients traditionally use port 5060 to connect to SIP servers  SIP acts as a carrier for the Session Description Protocol (SDP), which describes the media content of the session, e.g. what IP ports to use, the codec being used etc.  It is human readable and request-response structured  SIP messages: INVITE, ACK, BYE, REGISTER  SIP responses:  100 Trying  180 Ringing  200 OK  404 Not found  SIP shares many HTTP status codes, such as the familiar '404 not found'

8. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Protocols – H.323  H.323 is actually a family of protocols  H.323 ties together a number of protocols to allow multimedia transmissions over an unreliable packet based network  H.225 for call control and signaling  H.245 for exchanging terminal capabilities and creation of media channels  H.235 for security  RTP/RTCP for media

9. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Protocols – RTP (Real-time Transport Protocol)  Media applications are less sensitive to packet loss, but typically very sensitive to delays.  UDP is a better choice than TCP  RTP generally runs over UDP  RTP provides  payload-type identification  sequence numbering  timestamping  It does not guarantee any QoS

10. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Protocols - RTCP  Real-time transport control protocol (RTCP) is the counterpart of RTP that provides control services.  The primary function of RTCP is to provide feedback on the quality of the data distribution.  Statistics on a media connection  bytes sent  packets sent  lost packets  jitter  round trip delay  An application may use this information to increase the quality of service perhaps by limiting data sent or maybe using a low compression codec instead of a high compression codec  RTCP uses (RTP port + 1)

11. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Speech example Wel come to G S Lab

12. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Codecs  Convert speech to a digital format suitable to be transmitted over the network  Most codecs utilize compression to reduce the bandwidth requirement  But, heavy compression algorithms take time. This adds a delay to the conversation  Human speech is a very special signal and its characteristics are exploited in these algorithms

13. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Pulse Code Modulation  A PCM representation of an analog signal is generated by measuring ( sampling ) the magnitude of the analog signal at uniform intervals, and then quantizing it to a code.

14. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com G.711 (µ-law)  8000 samples per second  8 bits per sample  64 kbps  Logarithmic PCM (because the perceived loudness by humans is logarithmic)  µ-law: used in North America and Japan  a-law: used in the rest of the world

15. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Linear Predictive Coding  LPC starts with the assumption that a speech signal is produced by a buzzer at the end of a tube  The vocal tract (the throat and mouth) forms the tube, which is characterized by its resonances  The buzz is characterized by its intensity (gain) and frequency (pitch)  LPC analysis produces estimates for the pitch, gain and a set of numbers for the resonances  Voiced and Unvoiced

16. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com GSM codec  GSM uses linear predictive coding (LPC)  Speech is divided into 20 millisecond units (frames)  LPC parameters are determined for each frame  The number of bits needed to send these parameters is the bit-rate of the codec  For GSM, the bit rate is 13kbps

17. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Comparison between codecs CodecBit rateQuality (MOS) G.711640004.1 G.72980003.9 G.723.153003.6 LPC-1024002.7 Source for wave samples: http://www.signalogic.com/

18. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Network problems  Delay  Jitter  Echo  Congestion  Packet loss  Disordered packet arrivals

19. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Network issues - Delay  A delay of less than 150 ms is acceptable and usually goes unnoticed by humans  With delay greater than 400 ms, conversation starts becoming irritating  Coder delay is the time taken to compress a block of PCM samples  This delay varies with the codec used and processor speed  For G.729, delay is around 30ms  Packetization delay is the time taken to fill a packet payload with encoded speech  Queuing delay and Propagation delay at various network components  Jitter buffer delay

20. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Jitter  Variation in delay of packets is called Jitter  The effects of jitter can be mitigated by storing voice packets in a buffer upon arrival, before playing out  Increases delay by the length of the buffer

21. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Echo  Echo in telephony systems is caused by two main phenomena  Electrical echo due to imperfect impedance matching  Acoustic echo due to microphone pickup of audio output  Echo becomes noticeable only when there is a delay between speaking and hearing your voice echoed. (more than about 50 ms)  In PSTN calls, there is always echo, but it remains unnoticed because the delay is quite small  VoIP intrinsically has packetization, depacketization and processing delays built into its protocols  VoIP phones don't cause echo. They just make it audible by introducing an extra delay  Echo cancellation: Subtract from the received signal  Based on the response of the system to a short spike of sound  Echo cancellation is a hugely CPU-intensive process

22. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Advantages of VoIP  Reduction in costs  Uses the internet for long distance calls  Uses underutilized existing network capacity  Functionality  Especially for computer users – (click on name to call)  Merging of Data and Voice infrastructures  No need for separate cabling  Mobility  Wherever you are connected to the Internet, you can receive VoIP calls

23. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Disadvantages of VoIP  Quality  Due to low/variable bandwidth  Echo  Internet connection  VoIP usage is entirely dependent on the quality, reliability and speed of the internet connection  If the net is down, you have no telephony  Power  No phone calls in a power outage

24. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Services  Packet8, Vonage, Verizon  A black box with a phone attached  The user experience is almost indistinguishable from normal PSTN  The term “VoIP” is not used, instead – “Internet telephone” or “Digital telephone”  Broadvoice  Allow direct connect of SIP phones  Aimed at tech-savvy clients  Allows  Skype  Rely on the software client on the computer  Peer to peer  Routes calls through other Skype peers on the network  Proprietary, closed source

25. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Legal Issues  As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP in a manner similar to legacy PSTN services  In some countries, governments fearful for their state owned telephone services, have imposed restrictions on the use of VoIP  In India, it is legal to use VoIP. But it is illegal to have VoIP gateways inside India. This effectively means, people who have PCs can use it to make a VoIP call to any number. But if the remote side is a normal phone, the gateway that converts VoIP call to PSTN call should not be inside India

26. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com Cougar  What is it?  What can it do?  What software does it use?  How do I make calls?  Whom should I contact if I can’t?  Where to get more info?

27. Copyright © GS Lab 2006-7. All rights Reserved. http://www.gslab.com References  Wikipedia  http://www.linuxjournal.com/article/8424 http://www.linuxjournal.com/article/8424  http://www.cisco.com/warp/public/788/voip/delay-details.html http://www.cisco.com/warp/public/788/voip/delay-details.html  http://research.edm.uhasselt.be/jori/thesis/onlinethesis/chapter4.html http://research.edm.uhasselt.be/jori/thesis/onlinethesis/chapter4.html