VoIP Principles & Practice

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1 VoIP Principles & Practice Heison Chak SOMA Networks Inc. This tutorial is copyright by Heison Chak. It may not be used in whole or part for commercial purpose without the express written permission from Heison Chak. Heison Chak works for SOMA Networks as a network engineer. In this capacity, he focuses on network management and performance analysis as well as the implementation of data and voice networks. Presently he has undertaken to design a VoIP platform and migrate SOMA Networks to it from an existing legacy PBX system. Heison is an active member of the Asterisk community. Heison can be reached at Some of the materials are derived from columns and articles of Digium, Cisco, Intel and the voip-info wiki.

2 Tutorial Structure VoIP Principles PSTN, SS7 Telephone Numbering Plans VoIP vs IP Telephony VoIP Protocols and CODECs VoIP Quality VoIP Practice on Asterisk What is Asterisk? Asterisk by example Fun things to do with Asterisk Version APR 05 USENIX Heison Chak 2 Intended Audience: Managers and system administrators involved in the evaluation, design, implementation, and deployment of VoIP infrastructures. Participants do not need prior exposure to VoIP but should understand network principles. Attendees will come away from this tutorial with strategies for cost saving improvements to their existing infrastructures and practical information for deploying VoIP in a variety of environments. It does not cover: Step by step implementation (links and examples may be provided) Automatic Call Distribution (ACD) call queues and agents Analog Display Services Interface (ADSI) Media Gateway Control Protocol (MGCP) Skinny Client Control Protocol (SCCP)

3 Acronyms PSTN Public Switched Telephone Network POTS Plain Old Telephone Service TDM Time Division Multiplex DTMF Dual Tone Multi-Frequency SS7 Signaling System #7 ATA Analog Terminal Adapter SIP Session Initiation Protocol H.323 Video conferencing Protocol RTP Real Time Protocol IAX Inter Asterisk Exchange (pronounced eeks ) PBX Private Branch Exchange CODEC encoder/decoder MOS Mean Opinion Score Version APR 05 USENIX Heison Chak 3 PSTN the standard telephone service that most home use. Also referred to as Plain Old Telephone Service (POTS) TDM a scheme in which numerous signals are combined for transmission on a single communication channel DMTF used in touch tone dialing, a method where 2 distinct tones are sent for each digit dialed SS7 a global standard for telecommunications defined by ITU used in the PSTN ATA converts a POTS phone into an IP phone SIP a voice over IP protocol used for setting up communications sessions H.323 an ITU standard for real-time interactive voice and videoconferencing over LANs and the Internet RTP used to carry real-time data, such as audio, video over UDP IAX Asterisk proprietary VoIP protocol for SIP/H.323 replacement PBX private telephone network used within an enterprise CODEC converts analog signals into a digital format for transmission, and converts received digital signals back to analog format MOS scale commonly used in subjective voice quality tests

4 Public Switched Telephone Network (PSTN) Collection of interconnected systems operated by telcos and PTTs (postal, telephone, and telegraph) Started as human-operated plugboards Almost completely been made digital POTS (Plain Old Telephone Service) features FXO, FXS (Foreign Exchange Office & Station) Version APR 05 USENIX Heison Chak 4

5 PSTN (cont.) Version APR 05 USENIX Heison Chak 5 Two common interface to a standard POTS: Foreign Exchange Office (FXO) an interfaces that connect to a phone line, supply access to PSTN FXO interface use FXS signaling Foreign Exchange Station (FXS) an interface which drives a telephone, delivery battery and provide ringing FXS interfaces are signaled with FXO signaling

6 PSTN (cont.) Analogue signal is digitized at telco office 8kHz and 8 bits per sample (64kbps) DS0 Several DS0 are usually combined into a fatter stream US 24 DS0 channels combined into a T1 (DS1) Europe 32 DS0 are combined into a E1 Channels are separated, D/A conversion takes place and delivered to the receiving phone Version APR 05 USENIX Heison Chak 6 k = 1000 K = 1024 b = bits B = Bytes

7 PSTN (cont.) 24x DS0 combines to form DS1 TDM signaling is used to carry voice signals in time slots 193 Bits F Bit Time Slot 1 Time Slot 2 Time Slot 3 o o o o o o o o o o o Time Slot 23 Time Slot 24 8 Bits 8 Bits 8 Bits 8 Bits 8 Bits Transmission Rate = bps (8000 frames/sec * 193 bits/frame) DS1 is a protocol, T1 is a specific AT&T implementation of this protocol Version APR 05 USENIX Heison Chak 7

8 CLEC vs ILEC Local exchange carrier (LEC) provides voice-grade telecom service under regulation Incumbent LEC (ILEC) provides local services e.g. PacBell, Bell South Competitive LEC (CLEC) competes with ILEC e.g. Sprint, Primus Version APR 05 USENIX Heison Chak 8

9 Telecommunication Act of 1996 Signed by President Clinton on Feb 8, 1996 ILEC must allow CLEC to compete for local service Allow CLEC access to ILEC unbundled network elements (UNEs) e.g. local loop Making it easy for CLECs to compete without building new infrastructure Version APR 05 USENIX Heison Chak 9 Telecom act of 1996 is the first major overhaul of telecommunications law in almost 62 years, the goal of this law is to let anyone enter any communications business. Affecting telephony service (local and ld), cable programming and other video services, broadcast services and services provided to schools. In practice, ILEC work hard to make the competion diffcult The most important UNE for the CLEC is the local loop. With the local loop, CLEC will be able to connect their switches with the ILEC s switches and provide local services to end users

10 DTMF Dual Tone Multi-Frequency, system used by touch-tone telephones Telephone dialing Configure telephone exchange (switchboards) Voic applications Remote operation (home automation) Signal duration/operation: 40ms min Signal duration/non-operation: 23ms max Version APR 05 USENIX Heison Chak 10

11 DTMF (cont.) 16 distinct tones each generated by sum of two frequencies (low + high) Low/High 1209 Hz 1336 Hz 1477 Hz 1633 Hz 697 Hz A 770 Hz B 852 Hz C 941 Hz * 0 # D Version APR 05 USENIX Heison Chak 11 Your phone only uses 12 of the possible 16 tones, A thru D are system codes used to configure phone exchange The 2 key is row 0 (R1) and column 1 (C2), has frequency of = 2033 Hz The 9 key is row 2 (R3) and column 2 (C3), has frequency of = 2329 Hz

12 DTMF (cont.) 2 sine waves & the sum in time domain magnitude frequency response Version APR 05 USENIX Heison Chak 12 The higher of the 2 frequencies may have higher amplitude (be louder ) of 4dB max known as twist If lower frequency is louder, it is called negative twist Good DTMF decoders will not only detect the 2 received tones, but also examine the stop band and make sure they are under a certain threshold

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14 Signaling C.S. (cont.) SS7 signaling end points in a circuit switched network Version APR 05 USENIX Heison Chak 14 There are 3 kinds of signaling end points (SEP): Service Switching Point (SSP or central office switch) A telephone switch, or end node, equipped with SS7 software that sends signaling messages to other SSPs or STPs setup, manage, and release the voice circuits required to complete a call Signal Transfer Point (STP) A signal packet switch (or router) that receives and routes incoming signaling messages onward to their destinations. Service Control Point (SCP) An SSP may send a query message to a centralized database (an SCP) to determine how to route a call (e.g. a toll free 1-800/888 call in North America).

15 Signaling SS7 over IP Example of VoIP network configuration Version APR 05 USENIX Heison Chak 15 Media Gateway Terminates voice calls on inter-switch trunks from the PSTN, compresses and packetizes voice data and delivers compressed voice packets to the IP network Functions are performed in reverse order for voice calls originating in an IP network Media Gateway Controller Handles registration and management of resources at the media gateway(s) Exchanges ISUP messages with central office via a signaling gateway Media gateway controller is often called softswitch because vendors often use offthe-shelf computer platforms Signaling Gateway Provides transparent signaling between switched circuit and IP networks May terminate SS7 signaling or translate and relay messages over an IP network to a media gateway controller or another signaling gateway Often deployed in groups of 2 or more to ensure HA

16 Signaling (cont.) Telephone companies offload voice calls from PSTN to voice-over-internet Protocol Cheaper to carry voice traffic over IP networks than over circuit switched networks Computer replacing proprietary hardware IP telephony networks make better use of available bandwidth Version APR 05 USENIX Heison Chak 16 In a PSTN network, a dedicated 64 kilobits per second (kbps) end-to-end circuit is allocated for each call In a VoIP network, digitized voice data is highly compressed and carried in packets over IP networks Savings realized in using VoIP networks are often passed onto users in the form of lower costs

17 Telephone Numbering Plan 9 zones distinguished by general geographic region (e.g. +30 Greece, +39 Italy) has been defined by the ITU Inside regions, phone numbers contains area code, trunk prefix and subscriber number eg Version APR 05 USENIX Heison Chak 17 ITU stands for International Telecommunication Union Zone 1 North America Zone 2 Africa Zone 3, 4 Europe Zone 5 South/Latin America Zone 6 South Pacific/Oceana Zone 7 Russia and vicinity (former USSR) Zone 8 East Asia, Special Services Zone 9 West and South Asia, Middle East

18 Telephone Numbering Plan (cont.) Open numbering plans (7-dig) have different dialing arrangements for local and long distance calls Closed numbering plans (10-dig) use standard length numbers for all calls (a trend in many countries) Problem try to dial in SFO becomes (a local 415 number) Version APR 05 USENIX Heison Chak 18 Open numbering plans To call a number in San Francisco, dialing procedure will vary: xxx xxxx (local calls, no area code required) xxx xxxx (outside San Francisco) xxx xxxx (outside NANP) Closed Numbering Plans To call a number in Toronto, dialing procedure is: 416 xxx xxxx (local calls) xxx xxxx (outside Toronto) xxx xxxx (outside NANP)

19 NANPA North American Numbering Plan Administration Developed by AT&T in 1947 (impl. 1951) International Telecommunications Union (ITU) assigned country code 1 to NANP area NANP numbers are 10-digit numbers consisting of 3 digit numbering plan area (NPA), followed by 7 digit local number : NXX-NXX-XXXX where N=[2-9], X=[0-9] Version APR 05 USENIX Heison Chak 19 NANP includes United Stats and its territories, Canada, Bermuda, Anguilla, Antigua & Barbuda, the Bahamas, Barbados, the British Virgin Islands, the Cayman Islands, Dominica, The Dominican Republic, Grenada, Jamaica, Montserrat, St. Kitts and Nevis, St. Lucia, St. Vincent and the Grenadines, Trinidad and Tobago, and Turks & Caicos. ITU allocated a number for each country (e.g. European ), but it lacks a NANPA equivalence in Europe & Asia There are standards but doesn t seem to extend past the borders

20 NANPA (cont.) Easily recognizable code (ERC) are special area codes with identical 2 nd & 3 rd digits ERCs designate toll-free service e.g. 800, 877, 866, N11 ERCs not used as area codes, provide access to special services Version APR 05 USENIX Heison Chak 20 N11 code listing 211 community information and referral services (US) 311 non-emergency police and other governmental services (US) 411 local directory Assistance 511 traffic and transportation information (US); Reserved (Canada) 611 Repair service 711 Telecommunications Relay Service (TRS) 811 Business Office 911 Emergency

21 Agenda - VoIP Principles PSTN, SS7 Telephone Number Plans NEXT: VoIP & VoIP Protocols Version APR 05 USENIX Heison Chak 21

22 VoIP vs IP Telephony VoIP and IP Telephony (aka ToIP) are different VoIP refers to transferring digitized audio data across an IP network IP Telephony implies that this audio data is received by a Plain Old Telephone Service (POTS) telephone Version APR 05 USENIX Heison Chak 22 VoIP Voice quality according to applications need Connect members of the same organization IP Telephony Telephone call quality Connect anyone with necessary device with anyone on the planet with a phone

23 Benefits of VoIP - Providers Easier for ISPs to become a CLEC Go facility-less Rich call features Integration with other services Virtual PBX hosting Conference bridge Wake up call service Cost reduction Version APR 05 USENIX Heison Chak 23 While ISPs can become CLEC easier, differentiating themselves from competitions, there is significant cost savings to use integrate VoIP into existing data network

24 Benefits of VoIP Household Multi-ring, cascade ring with different technologies (aka follow me) Simultaneous calls on the same line Accurate CallerID (technically) Phone number portablility (except for mobile) Prevent phone spam! Better use of CPU Version APR 05 USENIX Heison Chak 24 Single number that rings home, office and cell phone only give out 1 number VoIP can carry multiple calls depending on your Internet bandwidth and choice of CODEC Voic / attachment, N-way calling, call forwarding, online call detail record viewing You may be able to migrate to VoIP without changing your telephone number

25 Benefits of VoIP - Business Reduce the number of circuits Bypass expensive access charges Save on taxes Reduce inevitable finger pointing between IT and service provider Toll-free numbers / call centers Version APR 05 USENIX Heison Chak 25

26 VoIP Service Provider (VSP) Provide bridge between VoIP and PSTN Deliver dial tone via broadband (Cable/DSL) Analog Terminal Adapter (ATA) turns POTS phone into an IP phone Monthly fee ~ $15-50/mo Pay as you go ~ as low as $0.02/min Telephone number portability Version APR 05 USENIX Heison Chak 26

27 VSP (cont.) Usability varies Availability, Power outage DSL vs. Cable NAT/firewall CODEC selection Delay, Echo, Jitter Privacy 900 numbers, Collect calls Version APR 05 USENIX Heison Chak 27

28 VSP (cont.) Tivo, alarm systems, point of sale Intercom systems within buildings Local telephone book listing Local number portability E911 and other essential services Almost a replacement of a land line... Version APR 05 USENIX Heison Chak is the emergency number for US & Canada, other countries have their reserved numbers European Union (EU) required members to 112 as the official emergency number in Dec Countries already had 3-digit numbers (e.g. UK 999), have continued to operate them in parallel with the required 112

29 VSP (cont.) Household Plans Primus Vonage Business Plans NuFone Telappliant Private Number Service Free World Dialup IAXtel Skype DingoTel Version APR 05 USENIX Heison Chak 29 Private Number Service Provide numbers in a private address space users can call other users of same service but not PSTN Support PC to PC, PC to IP Phone and, IP Phone to IP Phone Clear 800 toll free numbers (sorry, no 900) Other Service Providers

30 VSP (cont.) Free World Dialup (www.pulver.com/fwd) Session Initiation Protocol service since Nov, 2002 Works with hard and software IP phones 6 digit number accounts Used by 100's of thousands of SIP users Service provides toll free calling FWD can connect to other VoIP networks Version APR 05 USENIX Heison Chak 30

31 VSP (cont.) IAXTel (www.iaxtel.com) Inter Asterisk exchange service offered by Digium Gnophone GTK based IAX phone 700 area code (e.g ) termination primarily used by Asterisk developers service also clears toll free calls connectivity to VoicePulse and FWD Version APR 05 USENIX Heison Chak 31

32 VSP (cont.) Skype (www.skype.com) Proprietary peer-to-peer protocol (by KaZaA) uses voice technology from Global IP Sound intuitive user interface, easy to configure (cordless phone) supernode maintain presence information and communicate with other supernodes bizarre routing unwanted activities not compatible with other networks Version APR 05 USENIX Heison Chak 32

33 VSP (cont.) DingoTel P2P network connectivity to PSTN USB dongle for radio Voice Recognition Version APR 05 USENIX Heison Chak 33

34 VoIP Protocols VoIP application consists of Signaling call control (setup, monitor, teardown) SIP, MGCP, H.323, SCCP Capability exchange interoperability, require & permit SDP, H.225, H.245 Passing media audio stream RTP, RTCP Version APR 05 USENIX Heison Chak 34 Session Initation Protocol (SIP) - RFC2543 Most well known in VoIP Supported by various hardware Session Description Protocol (SDP) RFC2327 Media negotitation signaling used by SIP and MGCP Embeded in the payload H.323 (ITU standard) LAN-based video conferencing (defines how audiovisual conferencing data is transmitted) Enables multi-vendor interoperability (Enables users to participate in the same conference even though they are using different video videoconferencing applications) Inter Asterisk Exchange (IAX) - under development Proprietary protocol designed to work with Asterisk Media Gateway Control Protocol (MGCP) RFC2705 Similar to SIP, it also uses SDP for capability exchange and RTP for media Skinny Client Control Protocol (SCCP) Proprietary protocol used between Cisco Call Manager and IP phones Nothing to do with SS7 SCCP (Signalling Connection Control Part) Real Time Protocol (RTP) RFC1889 Designed to carry real-time data, such as audio, video or simulation data over multicast or unicast network services Provides payload type identification, timing construction and loss detection via sequence numbers and time stamps

35 VoIP Protocols SIP SIP proxy makes requests on behalf of other clients SIP Proxy Version APR 05 USENIX Heison Chak 35 Src Dest Protocol Info SIP/SDP Request: INVITE , with session description SIP Status 407 Proxy Authentication Required SIP Request: ACK SIP/SDP Request:: INVITE with session description SIP Status: 100 Trying SIP/SDP Status: 200 OK, with session description RTP Payload type=itu-t G.711 PCMU RTP EVENT Payload type=rtp Event, DTMF One RTP EVENT Payload type=rtp Event, DTMF Two RTP EVENT Payload type=rtp Event, DTMF Three RTP EVENT Payload type=rtp Event, DTMF Four 4 Invite/receive calls using SIP (Session Initiation Protocol) and SDP (Session Description Protocol) SIP Message contains Proxy Authentication parameters SDP Message contains CODEC selection and other capacity parameters SIP can use TCP too!! But most implementations are UDP only.

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37 VoIP Protocols H.323 H.323 gatekeeper controls endpoints (e.g. phones) and performs address translation (directory lookup) Call control signaling, authorization, management GNU gatekeeper does proxy/nat traversal and local call routing Use of a gatekeeper is optional Version APR 05 USENIX Heison Chak 37 3 main components of a H.323 call: Call control (setup, connect) Media control (capabilities exchange, open logical channel) Media (RTP stream) Source Destination Protocol Info TCP 1239 > 1720 [SYN] TCP 1720 > 1239 [SYN, ACK] TCP 1239 > 1720 [ACK] H CS: callproceeding H CS: alerting TCP 1239 > 1720 [ACK] H CS: connect H.245 TerminalCapabilitySet H.245 MasterSlaveDetermination H.245 OpenLogicalChannel (g711ulaw64k) H.245 OpenLogicalChannelAck RTP Payload type=itu-t G.711 PCMU

38 VoIP Protocols - RTP Real Time Protocol (RTP) RFC1889 Designed to carry real-time data (e.g. audio, video) over UDP (multicast/unicast) Does not ensure real-time delivery!!! RTP provides services such as payload type identification, sequence numbering, timestamping and delivery monitoring Version APR 05 USENIX Heison Chak 38 RFC1889 has been obseleted by RFC3550 List of currently assigned payload types RTP header compression only works for short-delay unicast connections on a single link RTP UDP ports assignment unclassified audio whiteboard video

39 VoIP Protocols RTP (cont.) Timestamp is used to place packets in correct timing order (increased by time covered ) T covered = packetization interval * sampling rate e.g. audio packets containing 20ms of audio sampled at 8kHz, timestamp increases by 160 Sequence number is used to detect losses (increment of one) Version APR 05 USENIX Heison Chak 39

40 VoIP and NAT Firewalls NAT (RFC1631) turns one IP address into another IP address Used to translate private IP (RFC1918) to public (Internet) IP NAT works in Layer 3 (IP Layer) NAT breaks if a Layer 3 address appears in Layers 4-7 (e.g. embedded IP) Version APR 05 USENIX Heison Chak 40 NAT enabled devices provides the ability to rewrite IP address and port in the datagrams as they pass in and out of the network many applications embed IP address at Layer 4 through 7 NATs generally do not modify Layer 4-7 address embedded within the IP payload, with the exception of a proxy (e.g. rtp and ftp proxy on a BSD firewall)

41 VoIP and NAT Firewalls (cont.) BBI Alice Firewall Bob Bob can t initiate calls to Alice Alice can get audio to Bob but not back (one way audio) Bob can t hangup the call properly Possible solutions: Alice needs to know that she is behind firewall Firewall needs to be smarter Version APR 05 USENIX Heison Chak 41 Typical problem with NAT: User A sends an invite to User B NAT translates the Layer 3 address, but not the Layer 5 (SIP/SDP) addresses User B receives the invite and responds back to the NAT address. The signaling gets completed (for example, 200 OK) User A sends RTP to User B s address port Problem: User B tries to send RTP to User A s address port but this fails since it can t route to User A the SDP address and port which did not receive the NAT - ONE WAY AUDIO (User A User B only) If User A hangs due to One-way Audio, the BYE is sent to User B correctly Problem: User B hangs up, the BYE won t get to User A because the header address did not receive the NAT. Leaving the state of User A to be up (hung) until User A hangs up

42 VoIP Protocols - IAX SIP and H.323 are just signaling methods RTP carries the actual payload (audio) IAX carries both signaling and payload IAX trunking puts packets for multiple calls in one UDP packet (saving IP, UDP headers) IAX is NAT friendly Version APR 05 USENIX Heison Chak 42 Inter-Asterisk Exchange protocol is used by Asterisk as an alternative to SIP, H.323, etc. when connecting to other devices that support IAX Unlike SIP - an IETF standard, IAX is not a standard at this time; white paper is available at It requires an IAX client to know absolutely nothing about the network that it is on to operate IAX2 header is only 4 / 10 bytes in size Only a single port needs to be opened to permit its use Port number = 5036 (IAX1) / 4569 (IAX2) Source Destination Protocol Info IAX2 IAX, source call #3, POKE IAX2 IAX, source call #9761, ACK IAX2 IAX, source call #9775, NEW IAX2 IAX, source call #9, AUTHREQ IAX2 IAX, source call #9775, AUTHREP IAX2 IAX, source call #9, ACCEPT IAX2 IAX, source call #9775, ACK IAX2 Voice, source call #9775, GSM compression IAX2 IAX, source call #9, ACK IAX2 Mini packet, source call #9775, GSM compression IAX2 DTMF, source call #9775, digit IAX2 DTMF, source call #9775, digit IAX2 DTMF, source call #9775, digit IAX2 DTMF, source call #9775, digit 4

43 Agenda -VoIP Principles PSTN, SS7 Telephone Numbering Plans VoIP vs IP Telephony VoIP Protocols NEXT: VoIP CODECs & Quality Version APR 05 USENIX Heison Chak 43

44 VoIP CODECs Voice signals must be converted to a digital signal for transmission over a digital network CODEC (encoder/decoder) is used vary in sound quality, bandwidth & processing requirement Waveform no knowledge (e.g. G.711) Source based on original signal (e.g. G.729, GSM) Version APR 05 USENIX Heison Chak 44 Knowing actual bandwidth before choosing CODEC is important Trade off between bit-rate and quality of the speech obtained at receiving end Speech signals are highly variable Speech signals are often contaminated and consist of repetitive patterns that can be eliminated thru compression (comm. Channels, background noise, room reverberation) ITU-T recommendations G-series (G.711, G.729) CODEC Bit Rate G kbps (MOS of 4.2) G kbps G kbps G kbps G kbps ilbc 15 kbps GSM 13 kbps G kbps k = 1000; bps = bits per second; 1 Byte = 8 bits

45 VoIP CODECs (cont.) G.711 (64kbps) no audible loss G711 µ-law for US and Japan G711 A-law for Europe and abroad G.729 (8kbps) Most widely used, industry proven Proprietary license required A & B are orthogonal Version APR 05 USENIX Heison Chak 45 PCM as a dynamic range of 13 bits, to make it octet friendly, it is a 16 bit G.729 A reduced complexity G.729 B silence supression native G.729 is rarely used

46 VoIP CODECs (cont.) ilbc (13kbps) Internet low bit-rate CODEC, designed for narrow band Computational complexity in range of G.729A Royalty free CODEC Version APR 05 USENIX Heison Chak 46

47 VoIP CODECs (cont.) GSM (13kbps) Global System for Mobile widely used in cellular network deployments Provides good quality speech (not as good as G.728) Runs easily in real time (G.728 requires dedicated DSP) Version APR 05 USENIX Heison Chak 47

48 VoIP CODECs (cont.) Speex (VBR) Patent-free audio compression designed for speech Variable bit-rate operation (2 to 44kbps) Packet loss concealment CPU intense operation Version APR 05 USENIX Heison Chak 48 Pronounced as speaks

49 VoIP CODECs (cont.) Most CODECs are optimized for speech, cannot guarantee to reproduce DTMF In-band vs out-band DTMF RFC 2833 defines RTP Payload for DTMF digits RTP events are sent separately and tones regenerated at receiving end Version APR 05 USENIX Heison Chak 49 Signals > 40ms will be sent as RTP events with duration RFC 2833 RTP Event Event ID: DTMF Five 5 (5) 0. = End of Event: False.0... = Reserved: False = Volume: 10 Event Duration: 160 Volume is the power level of the tone, expressed in dbm0 after dropping the sign [0, 36] dbm0 is valid DTMF < -55 dbm0 just be rejected Larger values denote lower volume Event duration is the total number of samples taken at 8kHz 160 samples = 20ms this field is sufficient to express event duration of up to approx. 8 secs

50 VoIP CODECs - packetization IP UDP RTP Payload depends how much audio fits here Packetization interval G.711 (64kbps) overhead vs. payload G.729 (8kbps) overhead vs. payload 10ms 80 bytes 1:2 10 bytes 4:1 20ms 160 bytes 1:4 20 bytes 4:2 30ms 240 bytes 1:6 30 bytes 4:3 Version APR 05 USENIX Heison Chak 50 Consider G.711 (64kbps): 8 8kHz sampling rate (8 bits * 8000 samples / sec = 64kbps) 10ms contains 80 samples (80*8 bits = 640 bits or 80 bytes) Now consider G.729 (8kbps) at 10ms: ( (overhead [40] + 10) bytes * 8 ) / 0.010s = 40kbps at the IP layer G.729 at 20ms: ( (overhead [40] + 20) bytes * 8 ) / 0.020s = 24kbps at the IP layer Only 1/8 of G.711 Payload is way bigger than overhead Increasing packetization to 30ms or higher introduce latency

51 VoIP CODECs packetization (cont.) t=0 10ms 20ms 30ms Packetization interval of 10ms vs 20ms Version APR 05 USENIX Heison Chak 51 Packetization concerns: Ability to put multiple calls in the same packets Savings in headers (Ethernet 16 bytes, IP 20 bytes, UDP 8 bytes & RTP bytes) Increasing packetization also Introduce latency Packet loss in channel

52 VoIP Quality ITU-T recommendation E.420 Connection establishment Connection retention Quality of the connection Billing integrity Let s focus on 2 quality metrics availability of service & quality of voice call Version APR 05 USENIX Heison Chak 52 ITU-T E.420 is one of the many E-series recommendations of the ITU-T (International Telecommunication Union Telecommunication Standardization Bureau) E-Series recommendations covers overall network operation, telephone service, service operation and human factors E.420 is responsible for checking the quality of international telephone service

53 VoIP Quality - Availability Availability (busy hours) and reliability (dropped calls, wrong numbers) Post Dial Delay (PDD) end of dialing until start of ringing Long PDD cause callers to give up and abandon the call Version APR 05 USENIX Heison Chak 53 PDD can be caused by: Call setup delay SIP has acceptable call setup delay, for simple calls H.323 delay worse at time, due to TCP

54 VoIP Quality Availability (cont.) Answer seizure ratio (ASR) number of calls ended with release cause 16, out of total number of call attempts in percent 60-70% in U.S. and European countries ASR include user behaviour e.g. dialing the wrong number In addition to ASR, network downtime is used to calculated service availability Version APR 05 USENIX Heison Chak 54 Release cause 1 = Unallocated number 16 = Normal call clearing (call answered and both parties hang up properly) 17 = User busy 18 = No user responding (telephone device not connected) 19 = No answer from User 21 = Call rejected 22 = Number Changed 27 = Destination out of order 38 = Network out of order 41 = Temporary failure In Canada, through Group Telecom, the hangup with PRI_CAUSE delivers these messages: PRI_CAUSE = 0,22,28,65-66 We re sorry your call cannot be completed as dialed, please contact the group telecom customer service center at if you require assistance PRI_CAUSE = 17,41 Slow Busy Signal (Normal busy signal you hear everyday) PRI_CAUSE = Ringing (Normal ringing you hear everyday) PRI_CAUSE = 42,54 We re sorry, all circuits are busy, please try your call again later PRI_CAUSE is an Asterisk variable that the Zap PRI channels examine and send as a PRI DISCONNECT message to the switch.

55 VoIP Quality Availability (cont.) % (or five-nine) is a mis-concept Bellcore originally defined 99.99% (53 mins/yr) VoIP networks should be greater than 99.94% to achieve equivalence with PSTN < 1 drop in 8,000 calls or 120 DPM dead air for >= 3 secs is considered a dropped call 50 ms recovery (for TDM) doesn't apply to VoIP < 5 ineffective in 10,000 calls or 500 DPM threshold of 30 secs Version APR 05 USENIX Heison Chak 55 For more details about service availability:

56 VoIP Quality Voice What contributes to poor voice quality? Cause Network congestion mis-ordered of arrival packets Link failures, routing instabilities Effect Packet loss, jitter delay/latency echo levels Version APR 05 USENIX Heison Chak 56 General Recommendation for utilization: Shared Ethernet (hubs/bridges) 30% 35 channels (10Mb/s, G ms) Half-Duplex Switched Ethernet 70% 825 channels (100Mb/s, G ms) Full-Duplex Switched Ethernet 80% 943 channels (100Mb/s, G ms)

57 VoIP Quality Voice (cont.) Measuring quality - Intrusive vs. non-intrusive Subjective testing yields MOS (Mean Opinion Score) Land-line 4.3, cellular PSQM (Perceptual Speech Quality Measure) Optimized for telephony speech signals, lacks VoIP PESQ (Perceptual Evaluation of Speech Quality) Designed to handle packet loss and jitter, however not designed for realtime apps Probe manufacturers - Minacom, IXIAcom, etc. Version APR 05 USENIX Heison Chak 57 Non-intrusive provides network characteristics, but not enough info to provides quality For reliable results, voice quality testing is mostly intrusive (well defined speech signal is inserted) Within the Absolute Category Rating (ACR) test method, the ITU five grade impairment scale is applied: Excellent 5 Good 4 Fair 3 Poor 2 Bad 1 Although testing is done without comparison to an undistorted reference; P.800 could be viewed as a comparison between a test signal and a reference in the mind

58 VoIP Quality Voice (cont.) Compute MOS value based on Delay and Loss Version APR 05 USENIX Heison Chak 58

59 VoIP Quality Voice (cont.) Measure quality of conversation to Boston & Los Angeles Reveals Round-Trip Latency, Packet Loss, Jitter, etc. Version APR 05 USENIX Heison Chak 59

60 VoIP Quality Voice (cont.) QoS introduces policies beyond best effort QoS enables acceptable and consistence call quality Packet Queuing FIFO (First In First Out) WFQ (Weighted Fair Queuing) CQ (Custom Queuing) PQ (Priority Queuing) CB-WFQ (Class Based WFQ) Traffic Shaping Version APR 05 USENIX Heison Chak 60

61 VoIP Quality Voice (cont.) Commercial VoIP QoS Solutions Open Source QoS implementations BSD - Alternate Queueing (ALTQ) Linux - Traffic Control (TC) Version APR 05 USENIX Heison Chak 61 /etc/altq.conf: # Strategy: Class Based Queue on external interface based on protocol # protocol: tcp (6), udp (17) # TOS: 0x08 (Throughput) # TOS: 0x10 (Low Delay) # ex0: external interface Interface ex0 bandwidth 640k cbq Class cbq ex0 root_class NULL priority 0 pbandwidth 100 Class cbq ex0 ctl_class root_class pbandwidth 5 control Class cbq ex0 def_class root_class pbandwidth 95 Class cbq ex0 voice def_class priority 3 pbandwidth 75 borrow filter ex0 voip tos 0x10 Class cbq ex0 file_transfers def_class priority 2 pbandwidth 10 borrow filter ex0 file_transfers tos 0x08 filter ex0 file_transfers tos 0x08 Class cbq ex0 other def_class priority 1 pbandwidth 10 borrow default

62 VoIP Quality Voice Jitter Irregular arrival of packets - Jitter Arrival delays may cause packets to be out of order Jitter buffer waits, reorders, releases packets Typical is 30ms to 50ms in depth Unnatural or robotic, silence or gaps in signal, clicking or popping sounds Watch out for out of sync audio and control signals Version APR 05 USENIX Heison Chak 62 Packets leaving source in order, likely to use different paths Jitter buffer introduces further delay and short silence during playback Jitter buffer must be kept to the right size Short enough to maintain natural pace Long enough to avoid packet drops

63 VoIP Quality Voice - Delay Algorithmic delay COder/DECoder Echo cancellation Network delay Transmission (network path, satellite path) Buffers Header compression Hardware delay Clock drift Version APR 05 USENIX Heison Chak 63 Clock drift affects jitter buffer (overrun & delay)

64 VoIP Quality Voice Delay (cont.) G.114 Guidelines < 150ms mostly acceptable > 400ms unacceptable (in general) Typical delays Terrestrial, national LD PSTN: < 50ms Terrestrial, international PSTN: ~ 100ms Cellular: mobile to PSTN: ~ 150ms Cellular: mobile to mobile: ~ ms Version APR 05 USENIX Heison Chak 64

65 VoIP Quality Voice Echo Leak-through of your own voice into our receive path Caused by reflector (electrical or acoustic), audible when > 25ms Usually a far-end (receiver) problem with analog device Echo cancellation examine Tx/Rx and removes accordingly (with DSP) Version APR 05 USENIX Heison Chak 65 Acoustic reflector Output from speakers bounced upon furnitures, walls, etc. and eventually makes it back to the mic Electrical reflector Hybrid (2 wire, 4 wire conversion) transformer out of phase Impedance mis-match Echo effect gets worse High amplitude Long delays Sources of Echo Foreign Exchange Office (FXO) Foreign Exchange Station (FXS) PSTN

66 Summary VoIP Principles Differences between PSTN and VoIP circuit switched vs packet switched high bandwidth vs low bandwidth dumb devices vs specialized computers ISUP signaling vs SIP and others to setup, tear down TDM signals in time slots vs RTP packets Version APR 05 USENIX Heison Chak 66 Businesses with extensive overseas contacts or high call volumes will benefit the most International companies and call centers have been reluctant due to previous investments in circuit-switched platforms

67 Summary VoIP Principles (cont.) NAT/firewall problems can be solved by: Protocol awareness on the firewall (proxy) Behind firewall awareness (embed external IP of firewall in Layer 4) Choice of CODEC Based on available bandwidth and usage High bit-rate handles packet drops better Low bit-rate have poor overhead to payload ratio Version APR 05 USENIX Heison Chak 67

68 Summary VoIP Principles (cont.) Audio Quality Echo is usually a far end problem, implement echo cancellation closest to the source Implementing QoS only guarantees circuit that you own Packetization intervals introduces delay, so does jitter buffer Keep your jitter buffer at the right size and watch out for clock drift, use NTP to prevent clock skew (time drift) Version APR 05 USENIX Heison Chak 68 When fully adopted, what happens when network goes down? QoS problem will improve over time

69 Summary VoIP Principles (cont.) Security challenge Eavesdropping / wiretap Balance your trade-offs Don t just measure savings, there might be other hidden costs Don t spend all your money on improving voice quality What if an ISP blocks your VoIP traffic? Version APR 05 USENIX Heison Chak 69

70 Agenda - VoIP Principles PSTN, SS7 Telephone Numbering Plans VoIP vs IP Telephony VoIP Protocols VoIP CODECs & Quality Summary NEXT: VoIP Practice on Asterisk Version APR 05 USENIX Heison Chak 70

71 What is PBX? Private Branch Exchange (PBX) is a private telephone network used within an enterprise Much less expensive than connecting external lines to phones Calling someone within the PBX is easier PBX system generally consists of: PBX switch Telephone wiring Handsets Version APR 05 USENIX Heison Chak 71 Users of the PBX share a certain number of outside lines for making telephone calls external to the PBX Telephone network may be as simple as in house Cat 3 wiring or as complex as multi-site ATM/Frame Relay networks PBX systems usually employs proprietary hardware & software + service contracts

72 * 8 # * 8 # Traditional PBX System Handsets PSTN Voic storage Music Source (Radio, CD Player) PBX Switch Featuring Conference Bridge Voic Music on Hold Version APR 05 USENIX Heison Chak 72

73 What is Asterisk? Software based PBX written in C Asterisk is fully Open Source Originally written by Mark Spencer Developed on GNU/Linux for x/86 Known to run on PPC, BSD & Mac OS X Digium is the primary developer and sponsor TDM400P (4 port FXO/FXS) X100P (1 port FXO) T100P (T1) TE400P (4 port T1) Version APR 05 USENIX Heison Chak 73

74 * 8 # * 8 # * 8 # * 8 # Conventional Softswitch Network IP Phones PSTN PSTN Gateway Conferencing Server POTS Gateway Voic Server Soft Switch Music on Hold Server Analog Phones Version APR 05 USENIX Heison Chak 74 Unless big bucks are paid for applications from the same vendor, integration will be a nightmare. It is sometimes slow for these big player to adopt new features Softswitch refers to technology that perform functions of a switch that provides similar services to a traditional circuit switch - using packet switching technology

75 * 8 # * 8 # * 8 # * 8 # Where Asterisk Fits IP Phones PSTN PSTN Gateway Conferencing Server POTS Gateway Voic Server Soft Switch Music on Hold Server Analog Phones Version APR 05 USENIX Heison Chak 75 There is a bunch of preferred hardware manufactured by Digium that makes integration with traditional TDM technologies very easily If you want to replace your Nortel option 11, you may find it difficult; since the handsets use a proprietary signaling. IP phones shown in the diagram can be SIP, MGCP, H323 and IAX

76 * 8 # * 8 # * 8 # * 8 # The Big Picture IP Phones PSTN PSTN Gateway Conferencing Server Voic Server POTS Gateway Music on Hold Server Soft Switch Analog Phones Version APR 05 USENIX Heison Chak 76

77 What is Asterisk? (cont.) Supports VoIP in these protocols H.323 SIP IAX Skinny MGCP And various CODECs GSM, G.711, G.726, SPEEX, ilbc, G.729, G.723, etc. Version APR 05 USENIX Heison Chak 77 Only G.723 pass-thru is supported due to licensing issue (http://www.dspg.com/technology/licensepricing.html) G.729 license can be acquired thru Digium at US$10.00 per channel (http://store.yahoo.com/asteriskpbx/asteriskg729.html)

78 What is Asterisk? (cont.) Key features of Asterisk Interactive Voice Response (IVR) Voic Conferencing Automatic Call Distribution (ACD) Call management/routing Standard call features Support VoIP and analog technologies Version APR 05 USENIX Heison Chak 78

79 Why Asterisk? Extreme cost reduction Low deployment cost Toll bypass Eliminates upgrade-path costs No per-seat licensing costs Take control of your phone system Highly customizable Deploy dynamic content via a telephone Free from vendor dependency Rapid and easy development environment Version APR 05 USENIX Heison Chak 79 Better use of CPU cycles when idle

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