Chapter 33 -- IP telephony (VoIP)

(Latest Revision: Sun Apr 29 15:00 2007 )

33.1 Introduction

33.2 The Motivation and Challenge of IP Telephony

Expectations are it will be cost-effective to use the Internet for both data and voice communication.
The basic idea:
- continuously sample audio
- convert to digital
- transmit across the Internet
- convert back to analog for playback
Details & Complications:
- can't wait to fill large packets ... transmission delay
- IP address <--> Telephone number translation
- call-acceptance protocol
- call-termination protocol
- bridging between Internet and the traditional Public Switched Telephone Network (PSTN)
- call forwarding
- call waiting
- voicemail
- conference calls
- caller ID
- PBX services

33.3 Encoding, Transmission, and Playback

Both the International Telecommunications Union (ITU) and the Internet Engineering Task Force (IETF) have created standards for IP telephony. Both ITU and IETF IP telephony standards agree on these points:
- use Pulse Code Modulation (PCM) for encoding audio,
- transfer digitized audio using the Real-Time Transport Protocol (RTP), and
- encapsulate each RTP message in a UDP datagram
TCP is not suitable for real-time voice transmission. It does no good to retransmit lost or late packets. RTP uses UDP and tolerates momentary silence when packets don't arrive in time for playback.
Each RTP message carries a sequence number and real-time clock value. Receivers use sequence numbers and real-time clock values to sequence playback correctly.

33.4 Signaling Systems and Protocols

Signaling consists of the processes of establishing and terminating a call.
There are proponents of both centralized signaling and distributed signaling for IP telephony.
Proponents of the centralized approach say it would help with providing service guarantees.
Proponents of the distributed approach feel it will be easier to implement and scale up.
Whatever signaling system is employed, it has to be compatible with Signaling System 7 (SS7), which is used by the traditional phone service.
Existing proposed IP signaling protocols:
- IETF's Session Initiation Protocol (SIP)
- ITU's H.323
- Megaco and Media Gateway Control Protocol (MGCP)

33.5 Basic IP Telephone System

An IP telephone can be a computer equipped with audio hardware and special software, or it can be a separate specialized hardware unit.
Besides IP telephones, an IP telephony system requires a Media Gateway Controller to allow telephones to "find each other."

33.6 Inter-Operation with Other Telephone Systems

An IP telephone system needs:
- a Media Gateway, and
- a Signaling Gateway
in order to interoperate with the Public Switched Telephone Network (PSTN) or another IP telephone system.
A Media Gateway provides translation between different audio formats.
A Signaling Gateway translates between different signaling protocols.
A Media Gateway Controller coordinates media and signaling gateways.

(See figure 33.2 below.)

33.7 Alternative Terminology and Concepts

33.7.1 Session Initiation Protocol (SIP) Terminology and Concepts
- A user agent makes or terminates phone calls.
- A location server manages database of info about each user & is contacted during call setup.
- A proxy server forwards requests, handles routing, & enforces policy.
- A redirect server handles call forwarding & 800-number connections.
- A registrar server updates the location database & authenticates registration requests.
33.7.2 H.323 Terminology and Concepts
- A terminal functions as IP telephone & perhaps can transmit data and/or video
- A gatekeeper performs location and signaling & coordinates operation of gateway.
- The gateway connects IP telephone system with the Public Switched Telephone Network (PSTN)
- A multipoint control unit (MCU) provides such services as multipoint conferencing.
33.7.3 ISC Terminology and Concepts
- The International Softswitch Consortium (ISC) presents a unifying model and defines a list of functions that suffice for all situations:
  - Media Gateway Controller Function (MGC-F)
  - Call Agent Function (CA-F)
  - InterWorking Function (IW-F)
  - Routing Function and Accounting Function (R-F/A-F)
  - Signaling Gateway Function (SG-F)
  - Access Gateway Signaling Function (AGS-F)
  - Application Server Function (AS-F)
  - Service Control Function (SC-F)
  - Media Gateway Function (MG-F)
  - Media Server Function (MS-F)

33.8 Proposed Protocols and Layering

The figure below depicts proposed protocols and their positions in the 5-layer Internet Reference Model.

33.9 H.323 Characteristics

33.10 H.323 Layering

H.323 can use both TCP and UDP over IP. For example audio over UDP and data over TCP.

33.11 SIP Characteristics and User Identification

33.12 SIP Methods

33.13 An Example SIP Session

33.14 Telephone Number Mapping and Routing

33.15 IP Telephones and Electrical Power

IEEE standard 802.3af provides low-voltage power over the wires used for ethernet.

33.16 Summary