1. VoIP
TDC 364

2. What is VoIP Used For? Reduced long-distance costs
Some cite this as a large business savings
Residential customers too
More calls with less bandwidth
New technologies allow voice to travel in less than 64 kbps channels (new voice compression techniques
Silence suppression

3. What Else is VoIP Used For?
More and better enhanced services
VoIP can be recorded, stored, processed, converted, etc. by the same hardware used for data
Computer telephony integration
Unified messaging
Most efficient use of IP
One common protocol

4. Four Additional Uses of VoIP
International calling
Telemarketing
PC and LAN dial one number after another
Worker reads from a script on their monitor
Depending upon answers/stored data, script changes dynamically
Telephone call goes through the pc

5. Four Additional Uses of VoIP
Call center
Telemarketing is outbound VoIP, call center is inbound VoIP
Automated attendant, automatic call distribution, interactive voice response
Call centers today are as dependent on the pc and LAN as they are on the telephone

6. Four Additional Uses of VoIP
Fax
The fax is not going away
Can be a legal document
Is tangible
Is by definition a copy of the original
Transcends languages and national borders
Millions of existing fax machines
But fax standards are antiquated
Fax over IP makes more sense

7. A Model for VoIP From business to business
Use: Faxing, tie-line replacement
Need: Better QoS for IP, managed IP network?
Outlook: Do it now
From business to residential
Use: Telemarketing
Need: IP-enabled PBX, ISP to PSTN gateways
Outlook: Do it carefully

8. A Model for VoIP From residential to business
Use: Call centers, catalog sales
Need: Voice-enabled Web site, IP-enabled ACD
Outlook: Do it carefully
From residential to residential
Use: Long distance replacement
Need: Many PSTN gateways, basic voice QoS
Outlook: Long distance now, local later?

9. What is the Basic VoIP Layout?
Voice
CODEC
Compression
Create voice datagram
Add header (RTP, UDP,
IP, etc)
Network

10. What is the Basic VoIP Layout?
Network
Process header
Re-sequence and
buffer delay
Decompression
CODEC
Voice

11. Traditional Network Characteristics
Voice
---
Short delay
Constant delay
No loss
No retransmission
Direct pass through
Data
Low error rate
Reasonable delay
Variable Delay
Packet Loss
Retransmission
Uses protocols

12. Packet Network Technologies
Same components, different performance
Internet – Routing (TCP/IP), frame relay, ATM
Intranet – Routing (TCP/IP), frame relay, ATM
Voice over networks
Internet – No goals, no guarantees
Intranet – Controlled environment, performance objectives, designed to perform

13. Voice Over Requirements
Compression
Reduced bps vs. quality
Silence suppression
Signaling
Echo control
QoS
Voice enhancements (calling features)

14. An Example: A Voice-Enabled Web Site
People talk on the telephone
People look at the web
What about voice and the web?
Visual orientation with human interaction
Flexible
Unlimited information
Wide availability (location and time)

15. Examples
Airline reservations (“Can I connect through Philadelphia instead?”)
Hotel reservations (“Does that room have a view of the ocean?”)
Ticket sales (“Can I get four seats together?”)
Stock trading (“Will I make the split requirements?”)

16. Call Center Without VoIP
3. Web site forwards
To call center
1. User clicks Web Call
2. User enter information
Call Me
Enter Information:
Name:
Account #:
Phone #:
Call information,
Account Information,
Etc.
5. User answers call,
Conversation begins
PSTN
4. Agent places PSTN call

17. Call Center With VoIP VoIP Call Internet Multimedia pc
3. Web site forwards all
Info to call center
1. User clicks Web VoIP Call
2. VoIP software uses same
IP connection to Web site
Call information,
Account information,
Etc.
VoIP Call
Internet
Multimedia pc
with VoIP software
4. Conversation through
VoIP software

18. The Web Added to the Call Center
PSTN
PBX/
ACD
Database
Voice network
to telephones
VoIP
Gateway
Agent with
telephone
and pc
Still two
networks
Agent with
telephone
and pc
Internet
Web
Server
Agent with
telephone
and pc

19. The Web Added to the Integrated Call Center
Database
PSTN
PBX/ACD
VoIP
Gateway
Agent with
telephone
and pc
Only one network
Agent with
telephone
and pc
Internet
Web
Server

20. The VoIP Gateway
The device that converts a traditional analog telephone call (voice and signals) into digital data that is sent over an IP network
Gateway functions include:
Destination lookup: converting a telephone number to an IP address
IP connection management: the use of protocols to establish, maintain, and teardown a call

21. The VoIP Gateway Gateway functions continued
Compression and digitization
IP packetization and transport
Advanced IP/PSTN signaling
Authorization, access, and accounting

22. The VoIP Gatekeeper An optional device, not required for H.323
Typically found in systems of significant size
Gatekeeper functions include
Address translation (supports the use of proprietary addressing schemes, such as mnemonics, nicknames, or address)
Admissions control (control the setup of VoIP calls between their terminals and gateways and the rest of the world; access granted or denied based on authentication, source or destination address, time of day, etc.; essentially a security mechanism)

23. The VoIP Gatekeeper More functions:
Bandwidth management (controls calls and the bandwidth of each channel)
Zone management (a zone is a combo gatekeeper, gateway, terminals, etc; gatekeeper controls calls within its zone)
Call signaling (may act as a signaling proxy for terminals it represents; or as an initial point of contact for callers)

24. VoIP Protocols
There are two basic sets of protocols for supporting VoIP:
ITU-T’s H.323
First issued in early 1996
IETF’s SIP (Session Initiation Protocol)
Introduced in 1998

25. VoIP Protocols continued
Interesting facts about the two protocols:
H.323 is named packet-based multimedia communications systems
H.323 originally designed for X.25 and ATM
SIP designed specifically for voice over the Internet by the people that should know the Internet the best
Let’s talk about H.323 first

26. H.323 Video Audio Control Data H.261 H.263 (video coding) G.711 G.722
Term. To
Gatekeeper
signaling
H.225
Call
signaling
H.245
T.120
(multipoint
data
transfer)
RTP
RTCP
RTP
RTCP
UDP
TCP IP

27. The Various Pieces – G.711
G.711 is the international standard for encoding telephone audio on an 64 kbps channel. It is a pulse code modulation (PCM) scheme operating at a 8 kHz sample rate, with 8 bits per sample, fully meeting ITU-T recommendations.

28. The Various Pieces – G.722
ITU-T G.722 is the benchmark coder for wideband speech coding quality. The speech signal is sampled at samples/second. G.722 can handle speech and audio signal bandwidth up to 7 kHz, compared to 3.6 kHz in narrow band speech coders. G.722 coder is based on the principle of Sub Band - Adaptive Differential Pulse Code Modulation (SB-ADPCM). The signal is split into two sub bands and samples from both bands are coded using ADPCM techniques. The system involving G.722 coder can be used to work in three modes 64, 56 and 48 kbit/s. The latter two modes will allow an auxiliary data channel of 8 and 16 kbit/s respectively, within the 64 kbit/s channel.

29. The Various Pieces – G.723
G is a speech compression algorithm standardized by ITU. G has dual coding rates at 5.3 and 6.3 kbps. The vocoders process signals with 30 ms frames and have a 7.5 ms look-ahead and low distortion while passing DTMF tones through. The input/output of this algorithm is 16 bit linear PCM samples.

30. The Various Pieces – G.728
ITU-T G.728 is low delay speech coder standard, for compressing toll quality speech (8000 samples/second). The typical application of this speech coder is in telephony over packet networks, especially voice over cable and VoIP. This is a very robust speech coder, with very good speech quality, comparable to 32 kbit/s ADPCM. G.728 coders are based on the principle of Low Delay-Code Excited Linear Prediction (LD-CELP).

31. The Various Pieces – G.729
G.729 is an 8 kbps Conjugate-Structure Algebraic-Code-Excited Linear Prediction (CS-ACELP) speech compression algorithm approved by ITU-T. G.729 Annex A is a reduced complexity version of the G.729 coder. G.729 AB speech coder was developed for use in multimedia simultaneous voice and data applications. The coder processes signals with 10 ms frames and has a 5 ms look-ahead which results in a total of 15 ms algorithmic delay. The input/output of this algorithm is 16 bit linear PCM samples.
Forward error correction (FEC) is incorporated in the algorithm to achieve noise immunity of the data stream by including control bits into it.

32. The Various Pieces – H.225
H.225 call signaling is used to set up connections between H.323 endpoints (terminals and gateways), over which the real-time data can be transported.
Call signaling involves the exchange of H.225 protocol messages over a reliable call-signaling channel. For example, H.225 protocol messages are carried over TCP in an IP–based H.323 network.

33. The Various Pieces – H.225
H.225 messages are exchanged between the endpoints if there is no gatekeeper in the H.323 network.
When a gatekeeper exists in the network, the H.225 messages are exchanged either directly between the endpoints or between the endpoints after being routed through the gatekeeper.
The first case is direct call signaling. The second case is called gatekeeper-routed call signaling. The method chosen is decided by the gatekeeper.

34. The Various Pieces – H.245
H.245 control signaling consists of the exchange of end-to-end H.245 capability messages between communicating H.323 endpoints.
The H.245 control messages are carried over H.245 control channels. The H.245 control channel is the logical channel 0 and is permanently open, unlike the media channels.
The messages carried include messages to exchange capabilities of terminals and to open and close logical channels.

35. RTP – Real-time Transport Protocol
Provides support for the transport of real-time data such as video and audio
Used in conjunction with RTCP to get feedback on quality of data transmission (next)
The Internet has unpredictable delay and jitter. To help alleviate these problems, RTP provides timestamping, sequence numbering, and other mechanisms.

36. RTP – Real-time Transport Protocol
Timestamps are created by the originator as the data is sampled. These timestamps are then used to play the data back at the same rate.
Since RTP is usually run over UDP, RTP adds a sequence number to all packets (some packets are broken into smaller packets, all with the same timestamp, thus the need for a sequence number)

37. RTP – Real-time Transport Protocol
Payload type identifier specifies the payload format as well as the encoding and compression schemes.
Source identification informs the receiver where the data is coming from (example – in an audio conference, a user can tell who is doing the talking)

38. RTCP – Real-time Control Protocol
In an RTP session, participants periodically send RTCP packets to convey feedback on quality of data delivery and information of membership.
Five types of RTCP packets defined:
Receiver Report
Sender Report
Source DEScription
BYE
APPlication specific functions

39. H.323 Call Stages Discovery and Registration (RAS) – This is who I am
Call Setup (RAS/H.225/Q.931) – This is who I want to call
Call Negotiation (H.245) – These are our capabilities

40. H.323 Call Stages
Media Channel Setup (H.245) – Let’s open an audio channel
Media Transport (RTP/RTCP) – Send audio datagrams
Call termination (H.245/H.225/RAS) – We are done

48. LAN Telephony (A Little More Detail)
PSTN
PSTN
Access
Gateway
Ethernet Phones
Ethernet
LAN
Analog Phones
Converter
Gateway
WAN or
Internet
Gatekeeper
IP Router
PC-based
Virtual Phones

49. SIP Session Initiation Protocol
An application layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants
These sessions include multimedia conferences, Internet telephone calls, and multimedia distribution

50. SIP Session Initiation Protocol
SIP has important features:
Scalability
Interoperability
Extensibility
Flexibility
Mobility

51. SIP Session Initiation Protocol
SIP first initiates a session
It can also modify and end a session
In order to initiate a session SIP has to first locate the user
After finding the user SIP delivers a description of the session in order to inform the user

52. SIP Session Initiation Protocol
SIP only conveys the descriptions of the session and doesn’t know anything about the session itself
Most common protocol to describe the session is the Session Description Protocol (SDP)
After locating the user and conveying the description of the session, SIP conveys the response of the user
The user can accept, reject, or forward the session.

53. SIP Session Initiation Protocol
If the session is accepted then an active session has been initiated
After initiation, SIP can also modify the session by sending a new description
SIP is based on the request-response paradigm

54. SIP Session Initiation Protocol
Some methods manage the sessions:
Invite: indicates that the user is invited to a session (session description also included)
Ack: to confirm a session establishment (via Invite)
Bye: terminates session
Cancel: cancel a pending Invite

55. SIP More methods to manage the sessions:
Options: used to query the server for its capabilities
Register: used to bind a permanent address to the current location of the user

56. SIP
To establish a session, the caller sends an Invite to the user with whom they want to talk
The user’s address has form
User responds to Invite with Ack and session is established.

57. SIP There are numerous response codes: Informational Success
100 Trying
180 Ringing
181 Call is being forwarded
Success
200 OK
Redirection
300 Multiple choices
301 Moved permanently
302 Moved temporarily

58. SIP More response codes: Client error Server failure Global failure
400 Bad request
401 Unauthorized
482 Loop detected
486 Busy here
Server failure
500 Server internal error
Global failure
Busy everywhere

59. SIP
The messages are not directly sent to the user - instead delivered to a proxy server
Proxy server responsible for routing and delivering messages to the called party
Proxy servers also relay call signaling

60. SIP There are several types of proxy servers:
Call-stateful: track call state and provide a lot of services, but are not fast
Transaction-stateful: track the request and responses but not the call state or session
Stateless: just receive requests, forward them, then forget them; fast but few services

61. SIP Redirect Servers SIP Registrars
Redirect the requestor to the other servers instead of forwarding them
Redirection is useful if a user moves or changes the provider
SIP Registrars
Accept the registration requests of the users

62. ENUM and E.164
SIP addresses are like addresses - both can be resolved by DNS
What if you only have a telephone number?
You need ENUM and E.164
ENUM is a protocol that resolves fully qualified telephone numbers to fully qualified domain name addresses using a DNS-based architecture
ENUM relies on E.164

63. ENUM and E.164 E.164 is an international telephone numbering plan
A fully qualified E.164 number is designated by a country code, an area or city code, and a phone number
ENUM allows users to access Internet-based services and resources from Internet-aware telephones, ordinary telephones connected to Internet gateways or proxy servers, and other Internet-connected devices where input is limited to numeric digits.

64. How Does ENUM Work?
Phone number is translated into a fully qualified E.164 number: (first 1 is North America, + means fully qualified)
All non-digits characters are removed:

65. How Does ENUM Work?
The order of digits are reversed: (Why? DNS names are structured from right to left.)
Dots are placed between each digit: (Why? Helps with administration)
Domain “e164.arpa” appended to end: e164.arpa

66. TRIP
TRIP (telephony routing over IP) servers maintain and exchange information on what gateways are available to establish calls to ranges of telephone numbers
TRIP allows multiple service providers to route calls through each other’s gateways

67. Example SIP Dialogue INVITE sip:bob@acme.com SIP/2.0 Via: SIP/2.0/UDP
alice_ws.radvision.com
From: Alice A.
To: Bob B.
Call-ID:
Cseq: 1 INVITE
Subject: Lunch today.
Content-Type: application/SDP

68. Example SIP Dialogue Content-Length: 182 v=0
o=Alice IN IP
s=Call from Alice.
c=IN IP4 alice_ws.radvision.com
m=audio 3456 RTP/AVP
Response Message would then follow