The Basics of Voice over the Internet Protocol Frank M. Groom, Ph.D. Professor of Information and Communication Sciences Ball State University

The Telephone Network

Circuit Switch Voice Message Path STP Call Establishment Circuit Switch Circuit 22 Circuit 7 DPC OPC CIC= 22 Initial Address Called # Calling # DPC OPC CIC= 7 Initial Address Called # Calling # DPC OPC CIC= 22 Initial Address Called # Calling # DPC OPC CIC= 7 Initial Address Called # Calling #

San Francisco Chicago NY Virginia ISP Carrier POP Carrier POP Carrier POP Carrier POP ISP Managed Private/ Peer Network

Access to the Telephone Network

Broadband Access to the Telephone Network

Creating Paths for IP Packets Across the Telephone Network

Creating a Path for Calls and Packets

The Standard Telephone Protocols

Video Equipment Audio Equipment User and System Data Applications System Control H.245 Control H.225 Call Control H.225 RAS Control T.120 and H.225 Data Transfer Video Codec H.261 and H.263 Audio Codec G.711, G.722, G.723, G.728 G.729 Multimedia Protocols

Standard Bodies 1.ITU – telephone standards supports the H.323 local network and conference standard. 2.IETF- Internet group supports the browser-like approach endorsed by the telephone companies.

CHARACTERISTICS OF VOICE AND IP TRAFFIC

Voice requires a Call-Setup Message to be transmitted first to notify the receiver and an Acknowledgement Message returned. Voice requires a regularity (minimum delay) of transmission. Voice only needs 8 Kbps bandwidth for each call.

Packet Switching LANS and Router-based networks are sized based upon the clustering of a sum of independent packets submitted in a bursty fashion. More efficient usage of trunk bandwidth is accomplished by sharing rather than determining the correct number and speed of links and ports. The trade-off is higher delay and delay variation due to queuing, blocking and congestion against a strategy of over-provisioning facilities.

64kb WAN ~214ms Serialization Delay 10mbps Ethernet Voice Packet 60 bytes Every 20ms Voice1500 bytes of DataVoice Voice Packet 60 bytes >214ms Every >214ms Voice Packet 60 bytes >214ms Every >214ms Voice1500 bytes of DataVoice 1500 bytes of DataVoice Large packets can cause buffer filling irregularities resulting in voice degradation Buffers to adjust for Jitter can accommodate some delay and delay variation The Problem of Mixing Data Packets with Voice Large Packets “Freeze Out” Voice

TYPE BYTE COUNT PACKET COUNT 300B- 1500B 1 Client request 200B 1 File transfer 50, , Print submit 2, , Internet request 60 B 1 COMMON PACKET VOLUMES

Result 1.Use small packets to set up a voice transfer path. 2.Use small packets to transfer voice content (73 Bytes). 3.Prioritize voice over data.

Some Standards used by VoIP

BASICS FOR TRANSMITTING VOICE IN IP PACKETS

KEY PARAMETERS FOR AN ACCEPTABLE VOICE NETWORK

(1) the maximum amount of delay experienced, (2) the amount of packets that are lost, and (3) the amount of variation or jitter in the arrival rates.

Inter-site Voice Connection Alternatives

VOIP OVERHEAD AND ITS EFFECTS Packet and cell-based networks require an overhead for addressing and other indicators which adds to each packet and comprises up to 10% of the total packet size.

General VoIP Connection Model

General Home Connection Model

ASSIGNING IP TELEPHONE ADDRESSES – DHCP

VOICE OVER INTERNET PROTOCOL MODELS FOR CONNECTION

IP PBX PSTN Public Internet Gateway VOIP Router IP Phone Private Internet ENTERPRISE CONNECTION OVER PUBLIC AND PRIVATE NETWORKS

Two-Site Enterprise VOIP

Multi-Site Enterprise VOIP

Residence Dial over RBOC Supplied VOIP Service

Residence DSL VOIP

Residence Cable VOIP

VOICE OVER IP TERMINALS SIP Phones H.323 Phones

VOICE OVER IP USING THE H.323 PROTOCOL

The H.323 family of protocols covers the functions of: 1. call signaling 2. transport of the various media types 3. system control 4. special specifications for conferencing including both point-to-point and multipoint conferencing.

To perform these functions, the family of H.323 specifications include: (a) the control specifications of H.245 and H.225 for signaling and control of transmission, (b) the video specifications including those for video compression H.261 and H.263 which are performed by a video codec and are required due to the large volume that would be transmitted if left in a raw form, and (c) the data specification of T.120. However, the most crucial H.323 specifications for employment with voice transmission under a VoIP process are those regarding the compression of audio G.711, 722, 723,728, and 729. This compression is performed as a function of the IP handset and is termed the audio codec.

The Multi-layered Header of an H.323 Packet

Dialing, Compression, and Header Addressing Layers of H.323

H.323 AUDIO CODING AND COMPRESSION

H.323 CALL SETUP SIGNALING AND MESSAGE FLOW

OPERATION OF GATEWAYS AND GATEKEEPERS IN H.323 NETWORK

Admission-to-the-network-requests (ARQ), admission confirmation (ARC) or admission rejection (ARJ)

H.323 Call Setup and Transfer Messages through Gateway and Gatekeeper

Three Zones Communicating by Means of Regional Area Gatekeepers

REAL TIME TRANSFER PROTOCOL

RTP provides timing and sequencing benefits, but at the cost of adding to the considerable UDP/IP header overhead To assist UDP for reliability enhancement purposes, RTP provides an additional 8-byte header to accompany the UDP 12-byte header which has already been added to the 20-byte IP header An additional algorithm, the RTP Compression (CRTP) algorithm, is sometimes employed to drop the total header to 2 bytes instead of the uncompressed 40 bytes..

RTP added to H.323’s Q.931 Signaling Sequence

VoIP Gateways and Gatekeepers Exchanging Info with the WAN

Universal Scheme for Connecting VoIP Traffic to Multiple Nets

SESSION INITIATION PROTOCOL (SIP) FOR VOIP TRANSMISSION

Basic SIP Overall Network and Services Architecture

SIP MESSAGES

Messages exchanged by client phones with servers &destination phones 1. Register- Each phone must register its existence, its parameters, and it’s burned-in MAC address (likely an Ethernet address). It is then assigned a telephone number and an IP address. 2. Invite- Each phone invites another phone to join a session and to exchange a conversation. 3. Acknowledge- Each phone receives back an acknowledgement when a calling session has been established and the destination device agrees to converse. 4. Bye- Each device issues a Bye message to hang-up and takes down the conversation session. 5. Cancel- Bother servers and user devices can issue a cancellation message to stop a request in progress

Sequence Issuance of Messages between Source and Destination Agents

SIP has four headers - one used for Requests, one for Responses, plus an Entity Header and a General Header. - the Request header field modifies the request command. - the Response header field enables servers to send response information back to the requester. - the Entity header field indicates information about the message in the body of the transmitted request/response message

Transmitted Frame with SIP, TCP, UDP and IP Headers

SIP spec covers the 3 core components of VoIP system. a) SIP first covers the application-level user agent and a server agent that can act on behalf or the user agent and receive and respond to the user agent requests. These agents exist in IP phones, IP servers, and gateway devices. b) SIP then specifies three types of network servers that act on behalf of clients to initiate, change, and terminate sessions. These are the Proxy servers, Redirecting servers, and Location and Registration servers c) SIP also provides for addressing in the traditional Internet URL fashion, such as with aliases of the fashion: or physical addresses such

SIP ADDRESSING AND OPERATION

USER AGENTS USING SIP PROXY SERVERS

Initiating a SIP connection over an IP Network

Destination Agent Acknowledging a SIP connection over the IP Network

Two-way Communication with SIP connection over IP Network

FLOW USING PROXY AND REDIRECT SERVERS

SIP Signaling Sequence and Operation

Sequence of SIP steps performed: 1. Registering with the Registration Server, 2. Requesting with an “Invite” of the proxy Server 3. Which then the Proxy Server asks the Location Server where the desired individual is located. 4. The Redirect Server then informs the sending requester at which URL address the desired individual is now located. 5. The Proxy server can now make a connection to the destination user for the caller, substituting a national public IP address for the private local IP address used by many businesses.

GATEWAYS AND GATEKEEPER PROTOCOLS

For simple Voice over IP connections, H.323 or SIP protocols are satisfactory

MEDIA GATEWAY CONTROL PROTOCOL- MGCP

The MGCP protocol supports gateways between a variety of networks. Among these connections are: a)Gateways for trunk connections between telephone networks and IP networks. b) Gateways interconnecting telephone networks and ATM networks. c) Gateways from a standard PBX to a switch interface and further to a voice carrying IP network. d) Gateways from a home devices or home network to a connection to the Internet primarily through an Internet Service Provider (ISP). e) Gateway from a business or residence to the public Internet by means of an analog modem and a dial-up connection through the Public Switched Telephone Network (PSTN). f) MGCP provides connection, signaling, and call control over the PSTN. g) MGCP allows for a division of the functionality with part to be provided by the Media Gateway and other parts to be provided by a central Media Controller placed out in the network. h) MGCP defines a means of handling signaling and session management for multimedia (voice, video, and data) conferencing.

The principle components of a MGCP process include: 1. the originating IP phones, 2. the Media Gateway itself, 3. and the Call Agent as a centralized assistant placed out in the IP network.

Call Agent Signaling to IP or PSTN Network

Call Agent Signaling for Traditional Telephone Transmission through IP Network

THE MGCP COMMANDS The MGCP protocol implements a set of commands that control the interfaces for the media gateway. These commands are structured as a set of origination commands and a required response to each command.

A Notification Request which asks the Media Gateway to look for activity arriving on a specific port from a specific end user terminal (IP Phone). The Media Gateway sends a Notify response to the Call Agent that originally made the request to inform when one of those events occur. The Terminal’s Call Agent sends a CreateConnection command to connect to a specific port on the Media Gateway. The Terminal can subsequently change any of the parameters of that connection with the issuance of a ModifyConnection command. The DelectConnect can be issued either by the terminal or the Media Gateway when a connection is no longer necessary or can’t be maintained. The status of endpoints, connections, and existing calls can be monitored by the Call Agent by issuing AuditEndpoint or AuditConnect commands. The RestartinProgress is a notification message sent to the terminal’s Call Agent indicating that the Media Gateway or a set of connected end points are being restarted or reconnected.

The Architecture of a National IP-based SIP-VoIP Service Network

Carrier SIP Service Local PSTN Enterprise IP Network IP Telephone Standard Telephone Enterprise IP Network Standard Telephone IP Telephone National IP Network Enterprise IP Network Standard Telephone IP Telephone

Conclusion 1.Somebody has to pay. 2.Somebody has to pay taxes. 3.Somebody has to pay for equipment.