Overview of Voice Over Internet Protocol
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1 Overview of Voice Over Internet Protocol Purva R. Rajkotia, Samsung Electronics November 4,2004
2 Overview of Voice Over Internet Protocol Presentation Outline History of VoIP What is VoIP? Components of VoIP VoIP Protocols Basic VoIP Call VoIP Standards Regulatory Environment Benefits of VoIP Challenges for VoIP VoIP on CDMA2000 Business Applications Benefits Technology Adoption 2/48
3 Some Interesting IP facts 50% of all enterprise telephone stations sold in US were IP based -In-Stat/MDR More than 30% of US and UK population will subscribe to VoIP service in the next 3 years -Mercer Management Group Worldwide revenue from the corporate Internet Protocol (IP) telephony business will grow from $ 1 billion by the end of this year to $ 5.5 billion in Radicati Group 3/48
4 History of VoIP VoIP came into existence as a result of work done by few hobbyists in Israel in the year 1995,when only PC-to-PC communication was in vogue. Later on during 1995, Vocaltec Inc. released Internet Phone software. The software used to compress the voice signal, convert it into voice packet and finally ship it over the internet. The sound quality was not even close to that of the standard equipment in use at that point of time. This attempt was the first IP phone that came into existence. VoIP has made great progress since then. 4/48
5 So, what is VoIP? Voice Over Internet Protocol (VoIP) is the assembly of voice into IP data. This data can be transmitted over an IP network to an addressable (IP address) destination. Voice calls look like data calls. VoIP calls are packet switched and analog calls are circuit switched. Packet switched is data that can be routed through different routes on a network to reach a destination. Circuit Switched is a connection where a physical path is dedicated between two end points. 5/48
6 So, what is VoIP? Analog CODEC NETWORK Digital COMPRESS PROCESS HEADER CREATE PACKET DECOMPRESS ADD HEADERS Digital NETWORK CODEC Analog 6/48
7 Components of VoIP 7/48
8 Components of VoIP The gateway converts the signals from the traditional telephony interfaces (POTS, T1/E1,ISDN) to VoIP. An IP phone is a terminal that has native VoIP support and can connect directly to an IP network. The server provides management and administrative functions to support the routing of calls across they network. In a system based on H.323, the server is known as a gatekeeper. In SIP/SDP, the server is a SIP server. In a system based on MEGACO or MGCP, the server is call agent. The IP network can be a private network, an Intranet or the Internet. 8/48
9 VoIP Protocol Stack PRESENTATION ADDRESSING AUDIO CODEC DTMF ADDRESSING RAS (H.225 ) DNS RTP/RT CP UNRELIABLE TRANSPORT (UDP) H.245 Q.931 (H.225) NETWORK (IP) RELIABLE TRANSPORT (TCP) DNS LINK PHYSICAL 9/48
10 VoIP Protocols Multicast IP The objective here is to send one packet and have it received at many destinations. In order to propagate these datagrams to multiple destinations, the routers within the network infrastructure operates with modified routing protocols. These multicast routing protocols construct logical spanning tree, which describes how the multicast traffic flows to the end stations. Real Time Transport protocol provides end-to-end delivery service for data that requires real time support, such as interactive audio and video. The services provided by RTP includes payload type identification, sequence numbering, time stamping and delivery monitoring. Real Time control Protocol monitors the quality of service and conveys information about the participants in an ongoing session. It is based on the periodic transmission of control packets to all participants in the session, using the same distribution mechanism as the data packets. 10/48
11 VoIP Protocols Resource Reservation Protocol ( RSVP) defines the QoS and the mechanisms to provide the QoS. RSVP is a control protocol, therefore it works in collaboration with-not instead of- traditional routing protocols. Session Description Protocol describes multimedia session for the purposes of session announcement, session invitation, and other forms of session related initiation. It conveys information about media streams in multimedia sessions to allow the recipients of a session description to participate in that session. Session Description Protocol information may be transported using SAP, SIP, RTSP, MIME or HTTP. SIP is a control or signaling protocol used for creating, modifying, and terminating sessions between participants. These sessions may include multimedia conferences, telephone calls, distance learning, or other types of multimedia distribution. SIP is a part of the internet multimedia protocol architecture which includes other protocols like RTP, SAP and SDP. 11/48
12 Basic VoIP Call Flow Station Initialization Connection Establishment Parameter Exchange Data Transfer Connection Termination 12/48
13 Basic VoIP Call 1.The caller picks up the handset. This signals an off-hook condition to the VoIP signaling. 2. The session application issues a dial tone and waits for the caller to dial a telephone number. 3. The caller dials the telephone number. The session application stores the dialed digits. 4. After enough digits are accumulated to match a configured destination pattern, the telephone number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to either the destination telephone number or a private branch exchange (PBX) that is responsible for completing the call to the configured destination pattern. 5. The session application runs the session protocol to establish transmission and reception channels for each direction over the IP net work. 13/48
14 Basic VoIP Call If Resource Reservation Protocol (RSVP) has been configured, RSVP reservations are put into effect to achieve the desired quality of service (QoS) over the IP network. 6. The coder-decoder compression schemes (codecs) are enabled for both ends of the connection using Real-Time Transport Protocol/ User Datagram Protocol/Internet Protocol (RTP/UDP/IP) as the protocol stack. 7. Any call-progress indications are cut through the voice path as soon as an end-to-end audio channel is established. Signaling carried over the IP network is encapsulated in Real-Time Transport Control Protocol (RTCP) using the RTCP application- defined (APP) extension mechanism. 8. When either person hangs up the phone, RSVP reservations are torn down (if RSVP is used) and the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger another call setup. 14/48
15 VoIP Standards Signaling ITU-T Standards and Recommendation H.323(Packet-based multimedia communication Systems) H (Call Signaling protocols and media stream packetization for packet based multimedia) H ( Annex G Gatekeeper to gatekeeper interdomain- communications) H.245 ( Control protocol for multimedia communications) H.235 ( Security and encryption for H- series multimedia terminals) H.450.x Supplementary services for multimedia 15/48
16 Signaling ( contd) H.323 Annex D Real time fax using T.38 H.323 Annex E Call connection over UDP VoIP Standards (contd) H.323 Annex F Single Use device T.120 series Data protocols for multimedia conferencing IETF RFCs and Drafts RFC 2543 SIP ( Session Initiation Protocol) RFC 2327 SDP ( Session Description Protocol) Internet Draft MGCP ( Media Gateway Control Protocol) Internet Draft MEGACO protocol RFC 1889 RTP, RTCP 16/48
17 Regulatory Environment CALEA and Law Enforcement Issue FBI wants the FCC to bring internet calling under provisions of the 1994 Communications of Law Enforcement Act ( CALEA), which requires phone carriers to provide them with the direct access to phone lines. But, in case of VoIP, the packet information travels in digital packets which is relatively easy to encrypt or to use secure tunnels making them inaccessible to law enforcement. 17/48
18 Benefits of VoIP VoIP has extremely efficient use of bandwidth as well as flexible technology. Hardware and protocols for VoIP are largely off the shelf, interchangeable and developed rapidly VoIP can also take advantage of the same functionality that is driving the internet, allowing providers to take advantage of the equipment at a higher level of productivity and cost savings. IP is the driving force for NGN( Next Generation Networks) with Convergence being an important requirement. VoIP is probably the simplest and easiest path to convergence. 18/48
19 Challenges for VoIP The current IP telephony services in the current state do not offer carrier grade standards. Worrisome is the fact that the callers addresses don t show up on emergency operators screens when they call 911. For VoIP on cable, the HSD (High Speed Data) connection is required, risking the caller to network and power outages. While the future is very promising, VoIP still has numerous technology challenges ahead, including integration, back office applications, OSS support, powering soft switches, return path and Packet cable. 19/48
20 VoIP on cdma2000 Service Option 60/61 to support the VoIP connection Allow Service Option 33 to set up the VoIP call Allow smaller packet sizes on the PHY layer to allow for fast er delivery of the VoIP packets Faster call set-up Header compression (1x: LLA-ROHC, Header Removal, HRPD- ROHC) QoS support ( QoS parameter negotiation and QoS scheduling) Segment based framing ( for HRPD). 20/48
21 Business Applications Benefits VoIP yields two types of benefits: - Soft Benefits Replacing a PBX with VoIP server may save a company specific amount of money every year. Soft benefits don t necessarily save money, or if they, do, they don t always save an easily calculated amount of money. But they do affect the overall bottom line in the future, like unified messaging etc. Hard Benefits Clearly defined cost savings. It is estimated that operator can reap cost savings by leveraging the investments made for the made for the same network to support HSD. If network powering along with the NIUs (Network Interface Units) and related functions and equipment are excluded then its possible to see considerable cost savings. 21/48
22 Technology Adoption Process Innovators Early Adopters Early Majority Late Majority Laggards Chasm Introduction Growth Maturity Decline 22/48
23 THANK YOU!!! 23
Receiving the IP packets Decoding of the packets Digital-to-analog conversion which reproduces the original voice stream
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