IP Telephony

IP Telephony

CONTENTS
Introduction Different types of IP Telephony What is h.323 H.323 components Protocol specified by H.323 H.323 call establishment Conclusion

IP Telephony

Introduction
Internet telephony uses the Internet protocol to send audio, video an data between two or more users in the real time. IP telephony is the integration and convergence of voice and data networks, services, and applications.

The main motivation of development of IP Telephony is the cost saving & integrating new services.
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Introduction (contd)
It is also called Voice Over Internet protocol (VoIP) , Internet telephony, Voice over broadband, broadband telephony.

IP Telephony

Introduction (contd)
Vocaltec introduced the first Internet telephony software product in early 1995. In 1996, Vocaltec with an Intel Company announced to produce the first IP telephony gateway.

IP Telephony

To make and receive calls with / without a computer Can surf the net while making calls Can make and receive call to / from PSTN Cost effective Digital features not commonly available on PSTN lines such as : voice mail caller ID conference and so on are available in VoIP.

Why VoIP?

PSTN
Voice networks use circuit switching. Dedicated path between calling and Called party. Bandwidth is reserved in advance. Cost is based on distance and time.

INTERNET
It uses packet switching. No dedicated path between sender and receiver. It acquires and releases bandwidth, as it is needed. Cost is not dependent on time and distance.

CIRCUIT SWITCHING

PACKET SWITCHING

IP Telephony is Different

IP

PSTN
Traditional Telephony: Smart switch Dumb phones
iLabs Voice Over IP Uing SIP 9

IP Telephony: Dumb network May 2005 Smart phones and servers

Different Types Of IP Telephony

PC to PC Phone-to-phone over IP PC-to-Phone Phone-to-PC

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PC to PC

Figure 1
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Phone-to-phone over IP using Gateways

Figure 2 (ref: www.iec.com)

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Phone-to-phone over IP using Adapter Boxes

Figure 3 (ref: www.iec.com)

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PC-to-Phone or Phone-to-PC

Figure 4 (ref: www.iec.com)

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IP Telephony

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Different type of IP telephony Standard

H.323 standard
Session initiation protocol (SIP) Media gateway to media controller protocol (MGCP)

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What is H.323
The H.323 standard for the transmission of real-time audio, video, and data communications over packetbased networks based on IP telephony. H.323 is a standard produced by the ITU-T Study Group 16. Currently the most widely-supported IP telephony signaling protocol.
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H.323 Components
Terminals Gateways Gatekeepers Multipoint Control Units (MCUs)
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IP Telephony

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H.323 Terminals
H.323 terminals are client endpoints that must support:
H.225 call control signaling. H.245 control channel signaling. RTP/RTCP protocols for media packets. Audio codecs.

Video codecs support is optional.

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Terminals
Telephones Video phones IVR devices Voicemail Systems Soft phones (e.g., NetMeeting)

Gateways
Gateway - composed of a Media Gateway Controller (MGC) and a Media Gateway (MG), which may coexist or exist separately The MGC handles call signaling and other non-media-related functions The MG handles the media Gateways interface H.323 to other networks, including the PSTN, H.320 systems, other H.323 networks (proxy), etc.

Gatekeeper
The Gatekeeper is an optional component It is used for admission control and address resolution It may allow calls to be placed directly between endpoints It may route the call signaling through itself to perform functions such as forward on busy, etc.

Gatekeepers
Address Translation Admission Control Bandwidth Control Zone Management Call-Control Signaling Call Authorization Call Management

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Multipoint Control Unit

MCU provide support for conferences of three or more endpoints. A MCU consist of:
Multipoint Controller (MC) provides control functions. Multipoint Processor (MP) receives and processes audio, video and/or data streams.

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Protocols Specified by H.323

Audio CODECs Video CODECs H.225 registration, admission, and status (RAS) H .225 call signaling H.245 control signaling Real-time transfer protocol (RTP) Real-time control protocol (RTCP)
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IP Telephony

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Audio CODEC It encodes the audio signal from the microphone for transmission on the transmitting H.323 terminal and decodes the received audio code that is sent to the speaker on the receiving H.323 terminal. Audio is the minimum service provided by the H.323 standard. ITU-T
G.711 (audio coding at 64 kbps),

G.722 (64, 56, and 48 kbps), G.723.1 (5.3 and 6.3 kbps), G.728 (16 kbps), and G.729 (8 kbps) recommendation are the audio 29 CODEC. IP Telephony

H.225 registration, admission, and status (RAS)

It is the protocol between endpoints and gatekeepers. It used to perform these tasks:
Gatekeeper discovery Endpoint registration
Endpoint location

Admission control

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H .225 call signaling

H.225 call signaling is used to set up connections between H.323 endpoints over which the real-time data can be transported. Two types : Direct call signaling Gateway routed call signaling

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Direct call signaling

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Gateway routed call signaling

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H.245 Control Signaling

Capabilities Exchange Logical Channel Signaling

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Real-Time Transport Protocol

Real-time transport protocol (RTP) provides end-to-end delivery services of real time audio and video. RTP is used to transport data via the user datagram protocol

Real-Time Control Protocol (RTCP)

RTCP provides control services for RTP.
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Process for Establishing Communication

Establishing communication using H.323 may occurs in five steps:
1. 2. 3. 4. 5. Call setup. Initial communication and capabilities exchange. Audio/video communication establishment. Call services. Call termination.

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Simplified H.323 Call Setup

Both endpoints have previously registered with the gatekeeper. Terminal A initiate the call to the gatekeeper. (RAS messages are exchanged). The gatekeeper provides information for Terminal A to contact Terminal B. Terminal A sends a SETUP message to Terminal B. Terminal B responds with a Call Proceeding message and also contacts the gatekeeper for perission. Terminal B sends a Alerting and Connect message. Terminal B and A exchange H.245 messages to determine master slave, terminal capabilities, and open logical channels. The two terminals establish RTP media paths.

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IP Telephony

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Versions of H.323
Version H.323 Version 1 H.323 Version 2 H.323 Version 3 Year May 1996 January 1998 September 1999

H.323 Version 4
H.323 Version 5 H.323 Version 6 H.323 Version 7
IP Telephony

November 2000
July 2003 June 2006 November 2009
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 Cheaper call rates Simplification High efficiency Calling person need not necessary to receive call. Better Voice Quality Using Wideband Codecs Adding new features and applications over time is easy. Integration of voice, data, fax, video is possible.

Challenges/Limitations
Some VoIP services dont work during power outages VoIP is susceptible to worms, viruses and hacking. Because VoIP uses an Internet connection, it's susceptible to all the hiccups normally associated with home broadband services. All of these factors affect call quality: Latency Jitter (time variation of a periodic signal or variable delay in communication) Packet loss Packet delay Phone conversations can become distorted, garbled or lost because of transmission errors.

 Packet Delay Packet Loss (no guarantee of delivering packets) Jitter (variable delay)

Qos- in VoIP
Quality of Service is essential for the success of VoIP. The human ear is extremely sensitive to even minor changes in an audio signal. Loss of quality occurs when the voice packets are transferred over the inherently unreliable packet-based networks.

Qos- in VoIP
Some methods to enhance the QoS achieved in VoIP are: Classification of Service: Using the TOS bits in the IP header to set a priority for the voice packet.

Tagging the packets with labels and using the labels to decide the route. Voice packets can be routed over less congested networks.
Reserving resources to meet requirements for bandwidth, delay, jitter, etc. along a particular path through a series of routers. Thus using above techniques the challenges/limitations of VoIP can be overcomed.

Conclusion
Internet telephony is cost savings and integrating new services. Internet Telephony is a powerful and economical communication options by combination of the telephone networks and data networks. Many public telecommunication operators are establishing their own IP telephony services, and using IP-based networks as alternative transmission platforms.
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 VoIP is predominately used for personal instead of enterprise-wide use.

The availability of high-quality audio using wideband codecs, video conferencing, and document sharing enables more effective and pleasant communication.

References :
Book on IP TelephonyOlivier Hersent, David Gurle & Jean-Pierre Petit.

www.iec.org/online/tutorials/ www.cis.ohio-state.edu/~jain/ www.cs.columbia.edu/~coms6181/ www.cisco.com/ www.tmcnet.com/ www.javvin.com/

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THANK YOU

24/08/2005

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SIP Endpoints are Intelligent

Endpoints are User Agents UA Client (originates calls) UA Server (listens for incoming calls) both SW and HW available

Zultys ZIP4x4

ZyXEL Prestige 2000W VoIP Wi-Fi Phone

ipDialog Siptone II

iLabs Voice Over IP Using SIP

Xten eyeBeam video softphone

Grandstream analog phone adapter (FXS gateway) 49

May 2005

 Voice over IP

contrast Introduction
Lower cost of network implementation Integration of voice and data applications New service features Reduced bandwidth

Replacing all traditional circuit-switched networks is not feasible. VoIP and circuit-switching networks coexist
Interoperation Seamless interworking

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VoIP (voice over IP - that is, voice delivered using the Internet Protocol) is a term used in IP telephony for a set of facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service IP Telephony 51

Understanding VoIP Signaling Protocols

Establishing VoIP Connections with H.323
H.323 networks contain three (3) primary solution components
Call processing servers store information on network topologies for routing calls
to VoIP gateways and end user devices Media gateways act as the H.323 termination endpoint and interface with nonH.323 networks, such as the PSTN (Public Switched Telephone Network) Gatekeepers function as the central unit for call admission control, bandwidth management, and call signaling
Gatekeepers separate call control and management functions from the gateways

Understanding VoIP Signaling Protocols

Establishing VoIP Connections with H.323

Source: Juniper Networks

 Terminals :
H.323 terminal can either be a personal computer or a stand-alone device, running an H.323 and the multimedia applications.

Gateways :
An H.323 gateway provides connectivity between an H.323 network and a non-H.323 network.

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Multipoint Control Units:

Responsible for managing multipoint conferences (two or more endpoints engaged in a conference) MCU contains
Multipoint Controller (MC) that manages the call signaling Multipoint Processors (MPs) to handle media mixing, switching, or other media processing

Multipoint Control Units:

MCUs provide support for conferences of three or more H.323 terminals. All terminals participating in the conference establish a connection with the MCU.

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Video CODEC It encodes video from the camera for transmission on the transmitting H.323 terminal and It decodes the received video code that is sent to the video display on the receiving H.323 terminal. The support of video CODECs is optional. ITU-T H.261 is the video CODEC recommendation
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H.323 Call Establishment

Figure 5 (ref: www.iec.com)

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H.323 Control signaling Flows

Figure 6 (ref: www.iec.com)

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H.323 Media stream & media control flow

Figure 7 (ref: www.iec.com)

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H.323 Call Release

Figure 8 (ref: www.iec.com)

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