Voice over IP Kiril Lashchiver SDBI 2005 Voice over IP sdbi 20052 Topics What & Why is VoIP Digital...

Voice over IP

Kiril Lashchiver

SDBI 2005

Voice over IP sdbi 2005 2

Topics

• What & Why is VoIP

• Digital PSTN

• Internet, IP/TCP/UDP

• VoIP


What & Why is VoIP


• Digital PSTN


• VoIP


What is VoIP?

The ability to make telephone calls and send facsimiles over IP-based data network with a suitable quality of service and a superior cost/benefit


In the beginning…

In 1995 VocalTec Communications Ltd. introduced the first internet phone

The voice was poor (due to high delay) but it was free innovative and most important it demonstrated VoIP capabilities and triggered new markets and studies


Why VoIP?

• Why another ‘telephone’ system?


Why VoIP (cont.)

• Cost Reduction

• Advanced Applications

• Standardization


Digital PSTN


• Digital PSTN


• VoIP


Digital PSTN

• PSTN: Public Switched Telephone Network• PBX: Private Branch eXchange• SSP: Service Switching Point


Nyquist theorem

• The Nyquist-Shannon sampling theorem is the fundamental theorem in the field of information theorem, in particular telecommunication

• The theorem states that:When sampling a band-limited signal the sampling frequency must be greater than twice the input signal bandwidth in order to be able

to reconstruct the original perfectly from the sampled version.


Nyquist example


Voice over PSTN

• The usable voice frequency band ranges from 300Hz to 3400Hz, or 0Hz to 4000Hz including guard bands

• According to Nyquist it would take 8000 samples per second to capture

4000(-epsilon)Hz perfectly• The signal is quantitized to 256 levels, 8 bits

per sample• 8000 samples * 8 bits = 64K bps


Voice in PSTN (cont.)

0

01100000

01000101

00110011

11100010

11100000

10100000

255

10110000

01011111

0.125 0.3750.250 1.000 1.1250.6250.500 0.750 0.875 Time

pitch


Voice over PSTN (cont.)

• Those 64Kbps streams are sent together into much larger streams – 30 voice channels – E1 – 2Mbps– 120 voice channels – E3 – 8 Mbps– 24 voice channels – T1 – 1.5 Mbps– 672 voice channels – T3 – 43 Mbps

…and that is just the beginning


Internet, IP/TCP/UDP


• Digital PSTN


• VoIP


Internet

Data

Packet 1 Packet 2

Each data chunk is broken into packets, attached withheader (and a sometimes a trailer)


Internet

1234


IP/TCP/UDP

• As we seen packet 3 didn’t reach it’s destination and packet 1 arrived last. The recovery (or lack of recovery) depends on the transport layer protocol

TCP (Transmission Control Protocol) UDP (User Datagram Protocol)

TCP will discover packet loss and ask the lost packet to be re-sent and the order will be corrected

UDP will not know about packet loss nor the order change.


IP/TCP

• TCP is a reliable but slow protocol, suits for web surfing or file transfer

• Adds big header

• Adds time delay

Therefore not suitable for real time applications


IP/UDP

• UDP is fast but not reliable protocol, suits for real time applications

• Sometimes used with RTP (Real Time Protocol) to add timing information to packets

• Small header• No time delay


Internet vs. PSTN

Size QoS Price

PSTN Worldwide (static)

Very Good High

Internet Worldwide (increasing)

Unknown Low


VoIP

• What & Why is VoIP• Digital PSTN• Internet, IP/TCP/UDP• VoIP

– QoS– Protocols– Security– News


VoIP QoS

• Providing a level of quality that at least equals the Digital PSTN

• QoS usually refers to exactness of the transmitted voice but can also be applied to network availability, telephone feature, scalability and security


VoIP Qos (cont.)

• However there are ‘few’ problems on the way– Delay– Packet Loss– Echo


VoIP - Delay

• Algorithmic delay

• IP+UDP+RTP = 40 bytes This typically added to voice packet that sent every 20ms (160 bytes)

Makes the total traffic 100kbps

Too much kbps. How we can improve?


Voice Coding

• Source waveformtransmission only the delta and the sign between n and n+1 samples, reduces to 32 – 16 kbps

• Vocoders (voice coders)attempt to model the vocal tract, very unnatural sound, low bit rate (e.g. 2.4)

• Hybrid (CELP) Combines the best of the two.

• Silence detection

Appendix A


Voice Coding (cont.)

• Performance of each Voice Coder is examined by the following:– Speech quality– Bit rate– Algoritmic delay– Complexity


Some examples

• PCM – 64 kbps

• ADPCM – 32 kbps

• GSM – 13.2 kbps

• CELP – 4.5 kbps

• OpenLPC – 1.4 kbps


VoIP – Delay (cont.)

• Processing delay:

• Caused by the process time of the decoder and the encoder

• Insignificant in good VoIP software


VoIP – Delay (cont.)

• Network Delay:

• Caused by the protocols used to transmit the voice

• Usually a function of the capacity of the links in the network.


VoIP – Packet Loss

• Internet – Packet switched

• Routers may drop packets

• Packets might get in the wrong order

• Packet loss of 10%+ is not acceptable


Packet Loss Compensation

• Skip the missed time and play the next packet

• Replay last voice packet received • Send redundant information (e.g. in nth

packet send also (n+1)th voice data• Hybrid, in nth packet send much lower

quality vocoder of the (n+1)th packet

0%: 10%: 20%: 50%:G.711:


VoIP – Echo Compensation

• Reflection of the speaker’s voice in speaker’s ear.

• Even in PSTN there is an echo, however due to a very short delay (less then 50 ms) it is not noticeable

• In VoIP the delay of the network almost always greater than 50 ms, thus echo cancellation techniques must be used


VoIP Protocols

• H.323– NetMeeting, GnomeMeeting and OpenH323

• SIP– Msn

• Others– Google talk– Skype– Many others…


H.323

• In May 1996 ITU released the H.323 specification, which defines how voice, data and video traffic would be transported over IP networks

• Written by telephony experts• Very complicated protocol• Rarely used in software VoIP phones, much

more popular in hardware(readable by machines)


H.323 (cont.)

• Allows different configurations of audio video and data

• Uses different protocols to manage different needs

• H.225 manages registration, admissions, status and call signaling


H.323Protocol


SIP

• Was developed by IETF and in November 2000 was accepted as signaling protocol for VoIP applications


SIP (cont.)

• Supports mobility• Lightweight protocol• Supports any media types (IM, voice, video)• Advanced features: changing setup during

the call, inviting participants during the call and call holding

• Easy to manage and troubleshoot


SIP protocol


SIP vs. H.323

• H.323 comes from the ITU (telephony) • SIP comes from the IETF and borrows many

concepts from Web, DNS and Internet e-mail• H.323 is a complete, integrated suite of protocols• SIP addresses only session initiation can be easily

combined with other protocols and services• H.323 being an umbrella standard, large and complex• SIP uses the KISS principle

Keep It Simple, Stupid


VoIP Security

• Both parts of VoIP call, the call setup messages and the voice stream need to be inspected by the firewall capable of both network and application level protection.

• Many different protocols

Maybe SIP???


Security – signaling path

• Problem: Signaling path sometimes contains vulnerable information about the users (phone numbers, account codes, id’s etc…)

• Call Hijacking• DoS

Partial Solution: authentication and integrity.


Security – Media stream

• Problem: Voice stream is vulnerable to eavesdropping and transport disruption

Partial Solution: The stream can be encrypted


Protocol Examples

googleTalk.cap

Msn.cap

Skype.cap

NetMeeting

SkypeGoogle Talk

MSN

netMeeting.cap


VoIP News

• Wireless VoIP• Large (and small) Companies change to

VoIP• Voice over ATM• The FBI encountered a new problem in the

war on terror: how to listen to internet phone calls


VoIP and IPv6

• As a real-time application VoIP should insure short arriving time

• As we seen high delay might be caused by the network

• VoIP packets are treated as ‘equal’ to other data packets

• IPv6 has a priority field, so urgent packets (like VoIP’s) will be forwarded first


IP Phone


Software IP phones


Anything missing? Let me know.


Appendix A How Vocoders Work?

• Over short intervals, about 2 to 40 milliseconds, human speech cam be modeled using three parameters– Sound source– Pitch – Recursive linear filter

• This model can be used to synthesize speech

Back


How Vocoders Work? (cont.)

• In order to improve the ‘robotic’ sound generated by the three methods the vocoder finds the best model to express the sound and adjusting the comparison, ignoring the differences that human ear can not hear


Bibliography

• Voice over IP, by Ayse Yasemin Seydim• Voice over IP, by Bur Goode, IEEE• VoIP and Security, by Greg S. Tucker• Speech synthesis using the CELP algorithm, by Geraldo LIno de

Campos• Secure IP Telephony, by Pingtel and Check Point• Simple VoIP Implementation, by Hong Xiong, Yuan Guo, Fang Zhu• VoIP, by Will Denis• It's Time To Take a Look At SIP, by Peter Morrissey• Computer Networking 3rd edition• RFC 781 IP• RFC 768 UDP• RFC 793 TCP• RFC 3261 SIP


Bibliography (cont.)

• http://en.wikipedia.org/wiki/Main_Page• http://www.nwc.com/1001/1001ws2.html• http://www.techonline.com/community/ed_resource/

feature_article/36563• http://www.hawksoft.com/hawkvoice/codecs.shtml• http://www.globalipsound.com/demo/index.php• http://www.slate.com/id/2095777/• http://www.networkitweek.co.uk/networkitweek/features/

2059672/rtfm-does-sip-work• http://www.ee.cityu.edu.hk/~cfchan/demo.html• http://www.protocols.com/pbook/VoIPFamily.htm• http://www.cisco.com/warp/public/788/voip/delay-details.html

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