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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 Metropolitan Network Carrier PoP National Carrier Backbone Network Carrier PoP Carrier PoP Carrier PoP Carrier PoP Carrier PoP Metropolitan Network Metropolitan Network Carrier PoP Metropolitan Network STP Call Establishment DPC 246-1-2 OPC 246-1-1 CIC= 22 Initial Address Called # Calling # DPC 246-1-2 OPC 246-1-1 CIC= 22 Initial Address Called # Calling # Circuit Switch DPC 246-1-3 OPC 246-1-2 CIC= 7 Initial Address Called # Calling # Circuit 22 Circuit Circuit 7 Switch Voice Message Path DPC 246-1-3 OPC 246-1-2 CIC= 7 Initial Address Called # Calling # Circuit Switch Chicago NY San Francisco Virginia Carrier POP Carrier POP Carrier POP Managed Private/ Peer Network ISP Carrier POP ISP ISP ISP Access to the Telephone Network Access to the Telephone Network Resi dence Customer Central Office Public and Private Internet IS P Business Customer Carrier POP Broadband Access to the Telephone Network Central Office IP Telephony Adapter PSTN DSL Modem Public and Private Internet Cable Modem Cable Head End Office Cable Modem Cable Head End Office DSL Modem Telephone Modem Dial-up Analog Line DS LAM Public and Private Internet Central Office IS P Public Metro Net Carrier POP PSTN IP IP PBX PBX IP Networks Private Line, WAN, Public Internet IP IP PBX PBX Creating Paths for IP Packets Across the Telephone Network Creating a Path for Calls and Packets MPLS Path Handler Call Agent Call Agent SS7 SS7 Voice Voice LOCAL PS TN PRI National IP Network MPLS Path Handler LOCAL PS TN PRI The S tandard Telephone Network Standard Telephone Protocols V.70, V.90, V92 Dial Terminals H.324 PC Terminal H.323 V0IP Network H.323 Gatekeeper Server S tandard Telephone IP Network IS DN Network H.320 Phone Digital Telephone H.323 Gateway Router H.323 Message Control Unit H.323 PC Terminal H.323 IP Telephone Multimedia Protocols Video Equipment Audio Equipment User and System Data Applications Video Codec H.261 and H.263 Audio Codec G.711, G.722, G.723, G.728 G.729 T.120 and H.225 Data Transfer H.245 Control System Control H.225 Call Control H.225 RAS Control 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. Voice Channel Voice Signal Output Voltage Frequency (K-Hertz) .2 1 Tone Dialing Signals 2 3 Systems Control Signals 4 Voice - the Most Restrictive and Smallest Tolerable Level of Delay CB Quality Satellite Quality FAX and Broadcast Quality Voice Quality 0 800 100 200 300 400 500 600 Time in msecs 150 msec Maximum Target Voice Delay 700 Packet Switching Packets Q PBX Q PBX 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. The Problem of Mixing Data Packets with Voice Large Packets “Freeze Out” Voice Voice Packet 60 bytes Every 20ms Voice Packet 60 bytes Every >214ms Voice Packet 60 bytes Every >214ms ~214ms Serialization Delay Voice 1500 bytes of Data Voice Voice 1500 bytes of Data Voice 10mbps Ethernet Voice 1500 bytes of Data Voice 10mbps Ethernet 64kb WAN Large packets can cause buffer filling irregularities resulting in voice degradation Buffers to adjust for Jitter can accommodate some delay and delay variation COMMON PACKET VOLUMES TYPE BYTE COUNT E-mail 300B- 1500B Client request 200B PACKET COUNT 1 1 File transfer 50,000-500,000 30 - 300 Print submit 2,500- 25,000 2 - 18 Internet request 60 B 1 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 Video Conferencing Standards NetType ISDN ATM PSTN POTS LANS Standard H.320 H.321 H.322 H.324 H323 Year Std 1990 1995 1995 1996 1996-98 Audio G.711 G.711 G.711 G.723.1 Codec G.722,28 G.722,28 G.722,28 G.72 G.722-29 Audio Rates 64kbps 64 kbps 64 kbps 6-8kbps 6-64kbp Video H.261 H.261 H.261 H.261 H.261 Codec H.263 H.263 H.263 H.263 Data T.120 T.120 T.120 T.120 T.120 Control H.230,42 H.242 H.242,30 H.245 H.245 H.225 Multiplexing H.221 H.221 H.221 Signaling Q.931 Q.931 Q.931 H.223 G.711 Q.921 H.323 PROTOCOL MULTIMEDIA OVER LANs VIDEO AUDIO CONTROL/ MANAGEMENT H.261 G.711, 722 RAS H.263 G.723 H.225 Signaling H.225 DATA Control H.245 T.124 G.728,29 RTP UDP X.224 Class 0 T.125 TCP T.123 IP LAN Layer 2 T.123 IEEE 802.3 BASICS FOR TRANSMITTING VOICE IN IP PACKETS 45 msec Sample Encode 64 msec Packetize < 100 msec Transmit Speaking Network Hearing Output 45 msec Decode 64 msec Jitter Buffer Reconstruct Receive Total of Less Than 250 msecs of Delay is Tolerable But Delay of Less than 150 msec is the Standard Voice Quality Delay From Various Compression Methods Delay Compression Method (msec) 64K PCM (G.711) 0.75 32K ADPCM (G.726) 1 16K LD-CELP (G.728) 3–5 8K CS-ACELP (G.729) 15 8K CS-ACELP (G.729a) 15 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 PBX PBX PSTN PBX Gateway Router Local S witch Edge Router WAN Services Public and Private Internet PBX Gateway Router Edge Router Local S witch 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 Telephone C.O. Switch Destination IP Telephone r S D m eri di a n IP Phone D S m e r id ia n D ua M o de M i r oc S Dua Mode Mi oc 1 2 ABC 3 DEF 4I 5 L 6M NO 7S 8 V 9 * 0Z # Y r ne a a e IP Server/PBX CISC OSYST EMS WR O Gatekeeper Cis co70 00 ERIES CISC OSYST EMS W WR O 1 2 3EF BC D A 4 I 5 L MN O 7 8 S V * 0 Z # r 6 9Y ne a a e Cis co 7 0 0 ERIES W LAN LAN IP Network LAN LAN Gateway Router Source PC Telephone Gateway Router Destination PC Telephone General Home Connection Model Central Office IP Telephony Adapter PSTN DSL Modem Public and Private Internet Cable Modem Cable Head End Office The Standard VOIP Packet Format VoIP Packet Link IP UDP RTP Voice Header Header Header Header Payload 12 Bytes X Bytes X Bytes 20 Bytes 8 Bytes G.711 Standard has a 160 Byte Voice Payload G.729 Standard has a 20 Byte Voice Payload G.723.1 Standard has a 24 Byte Voice Payload H.225 Call Setup H.323 Call Proceeding H.245 Invite SIP Acknowledge SDP CRC MGCP Acknowledgement SDP PSTN SS7 Signaling Network ASSIGNING IP TELEPHONE ADDRESSES – DHCP VOICE OVER INTERNET PROTOCOL MODELS FOR CONNECTION ENTERPRISE CONNECTION OVER PUBLIC AND PRIVATE NETWORKS IP Phone PSTN Public Internet IP PBX Gateway VOIP Router Private Internet Two-Site Enterprise VOIP IP Phone PSTN IP Phone IP PBX Public Internet IP PBX Gateway VOIP Router Gateway VOIP Router Private Line or Private Internet Multi-Site Enterprise VOIP IP Phone PSTN IP Phone IP PBX Public Internet IP PB X Gateway VOIP Router IP Phone Gateway VOIP Router Private Internet IP PB X Residence Dial over RBOC Supplied VOIP Service Telephone Central Office Building PSTN Public Internet Dialup Modem Gateway MultiService VOIP Switch Router Private Internet Residence DSL VOIP Telephone Central Office Building PSTN DSL Mode m DSLAM Gateway VOIP Router Public Internet Residence Cable VOIP PSTN Cable Head End Office Cable Modem Public Internet CMTS Gateway VOIP Router Private Internet 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. VoIP—Uses Audio Component of H.323 System Control and User Interface Video Video I/O Equipment Audio Audio I/O Equipment Video Codec H.261, H263 Audio Codec G.711, G.722, G.723, G.723.1, G.728, G.729 System Control H.245 Control Call Control H.225. 0 RAS Control H.225. 0 Receive Path Delay H.225.0 Layer LAN Stack User User Data Data Applications T.120 Session Layer and Above The Multi-layered Header of an H.323 Packet Reliable TCP Delivery Unreliable UDP Delivery H.245 H.225 Audio/Video Streams Call Control RAS TCP RTCP and RTP UDP IP Media Payload Dialing, Compression, and Header Addressing Layers of H.323 H.323 VoIP Model Signaling and Transport Caller E.164 Phone # Audio Codec (G.711, G729, G.723.1) H.225, H.245, RTP, RTCP UDP Port # IP Address MAC Address 802.3, ATM VPI, VCI Frame Relay DLCI Physical V.35, T1, DS-3 H.323 AUDIO CODING AND COMPRESSION H.323 CALL SETUP SIGNALING AND MESSAGE FLOW H.323 Call Setup Signaling Admission Request Admission Confirm H.323 Gateway RAS H.323 Gateway Setup Connect H.225 (Q.931) Capabilities Exchange Open Logical Channel H.245 Open Logical Channel Acknowledge Path Resv RTP Stream RTP Stream RTCP Stream Gatekeeper RSVP Media OPERATION OF GATEWAYS AND GATEKEEPERS IN H.323 NETWORK Gatekeeper Dua Mo de M i oc RRQ, RCF ARQ ARQ ACF RRQ, RCF H.225Call Set up Gateway ACF H.225Call Set up Response Gateway H.245 Call Management Real Time Message Transfer Flow Admission-to-the-network-requests (ARQ), admission confirmation (ARC) or admission rejection (ARJ) H.323 Call Setup and Transfer Messages through Gateway and Gatekeeper Gatekeeper Dua Mo de M i oc RRQ, RCF ARQ ARQ ACF RRQ, RCF H.225Call Set up Gateway ACF H.225Call Set up Response H.245 Call Management Real Time Message Transfer Flow Gateway Three Zones Communicating by Means of Regional Area Gatekeepers Gatekeeper 2 Gatekeeper 1 Gatekeeper 3 Dua Mo de M i oc D uaM o de M i oc D ua Mo de M i oc Gateway 1 Zone 1 Gateway 2 Zone 2 Gateway 3 Zone 3 REAL TIME TRANSFER PROTOCOL RTP Header V2 P X CC M PT Ti mestamp Sequence Number Synchronization Source Identifier UDP Header Source Port Destinati on Port Checksum Length Version IHL Type of Service Identification IP Header Ti me to Li ve Total Length Fl ags Protocol Fr agment Offset Header Checksum Source Address Destinati on Address Opti ons Paddi ng 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. 20 bytes IP Header 8 bytes 12 bytes 20 to 160 bytes UDP Header RTP Header Voice or Media Payload 40 byte Header Compressed to 2 Byte Header Compressed Header 2 to 4 bytes Voice or Media Payload 20 to 160 bytes RTP added to H.323’s Q.931 Signaling Sequence Realtime Transfer Protocol Contribution to H.323 Q.931Signaling H.323 Gateway Messages H.323 Gateway Messages TCP Connection Setup Alert Connect H.245 Address H.245 Gatekeeper Messages Open Logical Channel Open Logical Channel Acknowledge Q.931 Signaling Over TCP Gatekeeper Messages H.245 Over TCP Realtime Transfer Protocol Contribution RTP Stream RTP Stream RTP Stream RTCP Stream RTP Timed Streaming Voice Media Transfer over UDP VoIP Gateways and Gatekeepers Exchanging Info with the WAN Gatekeeper DuaM o de Mi oc Registration Request Gateway RCF Registration Confirm Voice Communication Packets Address Exchange WAN Network RRQ Voice Communication Packets Gateway Universal Scheme for Connecting VoIP Traffic to Multiple Nets National VoIP Gatekeeper Public and Private Internets Gateway Gatekeeper Router Local VOIP Network Gateway H.323 Over Local IP and Ethernet Network IP Phones Public Circuit Switched Telephone Network PBX Standard Analog and Digital Telephones SESSION INITIATION PROTOCOL (SIP) FOR VOIP TRANSMISSION Basic SIP Overall Network and Services Architecture Application Services SIP Servers SIP Servers SIP Proxy, Locate, Register, Redirect Processes SIP SIP SIP PSTN SIP RTP/UDP PBX IP Device w ith SIP Agents 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 Server Calling IP Phone Invite Invite Code 100 Try SIP Signaling (TCP) Code 100 Try Code 180 Ring Code 180 Ring Code 200 OK Code 200 OK ACK Media Transfer UDP and RTP ACK Media Transfer RTP Stream Called IP Phone 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 Gopher Kerb SMTP Telnet FTP SIP SNMP RPC UDP TCP IP Local Area or Wide Area Net Interface D MAGEL A N D MAGEL AN User Agent Server User Agent Client SIP Servers Proxy Redirect Location Register User Agent Server User Agent Client 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: abjones @bsu.edu or physical addresses such [email protected] SIP ADDRESSING AND OPERATION USER AGENTS USING SIP PROXY SERVERS Initiating a SIP connection over an IP Network Proxy Server Invite Redirect Server Invite Client Client IP-Based Network User Agent User Agent Destination Agent Acknowledging a SIP connection over the IP Network Proxy Server Response Code 200 Client Redirect Server Response Code 200 indicating Success Client IP-Based Network User Agent User Agent Two-way Communication with SIP connection over IP Network Proxy Server ACK User Agent Redirect Server ACK Acknowledgement RTP Voice Transfer User Agent IP-based Network Client Client FLOW USING PROXY AND REDIRECT SERVERS SIP Signaling Sequence and Operation SIP Server Calling IP Phone Invite Invite Code 100 Try SIP Signaling (TCP ) Code 100 Try Code 180 Ring Code 200 OK Code 180 Ring Code 200 OK ACK Media Transfer UDP and RTP ACK RTP Stream Called IP Phone 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. SIP Servers and Offered Services 1. Register “I am Frank Groom” Redirect User Locations Register SIP Proxy Server 3. Locate “Where is Kevin Groom at 555-9999?” 5. Proxy “I’ll establish the connection for you” 2. Invite “I w ant to talk to another User Agent” IP Devices w ith SIP Agents Frank’s IP Phone 4. Relocate “Kevin is now at [email protected]” SIP RTP/UDP Kevin’s IP Phone PBX 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. MGCP Call Agent MGCP Messages MGCP Messages IP NETWORK Media Gatway Media Gatway Call Agent Signaling to IP or PSTN Network SS7 Signal Call Agent PSTN MGCP Media Gateway MGCP Call Setup Signaling IP Network Media Gateway Voice Messages Media Gateway Call Agent Signaling for Traditional Telephone Transmission through IP Network Call Agent SS7 Signaling Network SS7 Signaling Network SS7 Signal Trunk Traditional Telephone Dial Link Media Gateway MGCP Call Setup Signaling IP Network Voice Messages Trunk Media Gateway Traditional Telephone Dial Link 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 Standard Telephone Local PSTN IP Telephone IP Telephone Enterprise IP Network Enterprise IP Network Local PSTN Standard Telephone Carrier SIP Service National IP Network IP Telephone Enterprise IP Network Local PSTN Standard Telephone Company AT&T SBC Price Service Availability n/a Local and Long Distance 100 US Mkts 2004 $29-40/ mo Local and Long Distance 100 cities 2004. Verizon n/a Bus and Res DSL, Local and LD Initial offer 2004 Bell South n/a Bus. Serv ice only annou nced 2004 Qwest n/a 14 State Svc Area, Local, LD 2004 Time Warner $39.95/ mo. Partner M CI/Sprint, local, LD begin 2004 MCI n/a TWC partner svc Announced 2004-2005 Sprint n/a TWC partner svc announced 2004-2005 Covad $35-60/ mo. Vonage $14.99-34.99 i2 Teleco m $9.95/ mo. National Bus/Res Local and LD Local and LD Reseller Program announced 4th qtr 2004 300,000 cust. 2004 2004 Galaxy Vo ice $19.95-34.95 Bus and Res cust, Local and LD 2004 8x8 2004 $20/ mo. Res and Small Bus., Local, LD Conclusion 1.Somebody has to pay. 2.Somebody has to pay taxes. 3.Somebody has to pay for equipment.