Voice over IP: Issues and Protocols

Voice over IP: Issues and Protocols Dhigha D Sekaran Sunil Chintakindi

Voice over IP: An Introduction Voice over IP is a technology allowing an enterprise to carry voice traffic over an IP data network at substantially reduced bandwidth. It is a a set of facilities for managing the delivery of multimedia information (including fax transmission) in digital form in discrete packets

Why VoIP ? Cost reduction for Voice Traffic Convergence of Voice and Data leverages infrastructure investment Big cost reduction for intra-site fax traffic Enables new multimedia applications to increase productivity VoIP can save you real money. Example 1: No network in place, install IP network between 4 sites, 1500 call minutes anywhere to anywhere is breakeven (all domestic, 0.10/min) Example 2: Network in place between 4 sites, voice goes for free, payback period << 1 yr Example 3: International Long Distance, savings are tremendous. Toll reduction Not only site to site, use packet network to bypass long distance network, and place calls from local point of presence.

Why IP ? IP in conjunction with TCP , is one of the most widely used networking technology made popular by Unix/Windows OS and the Internet According to estimates 90% of the corporate world will be using TCP/IP by 2K!! Any network that connects to the internet must run TCP/IP and we know Internet is expanding rapidly … . VoIP seems to be a very attractive technology!!!!

A Simple VoIP Network… IP Network PBX PC Phone VoIP Gateway Server VoIP Gateway Frame Relay or ATM Link LAN Fax

Standards and Protocols The Two Important VoIP protocols are SIP-Session Initiation Protocol H.323

Benefits of VoIP…. Low Cost of public Internet Long distance call for the price of local PSTN Integration of voice and data applications Internet–aware telephones Intranet Telephony Trivial implementation of existing services Video conferencing Voice traffic on data networks- 40% reduction in costs

Issues in VoIP Delays Congestion Jitter Packet Loss Limited Bandwidth Echo Interoperability Scalability

Issues in VoIP………….contd Delay Delays can be defined as the time taken for the voice packet to reach destination from source Human ear can feel delays greater than 250 millisecond Delays consists of Analog to Digital encoding( at sender side) Digital to Analog decoding( at receiver side) Processing at gateway routers Network –the physical medium Buffer(jitter)- packets are sequenced and buffered at the receiver side Can be minimized by giving priority to voice packets. Routers can be configured to do this Prioritization protocols like RSVP

Issues in VoIP………….contd Congestion Occurs when there are too many packets in the network Results in delay Minimized by Prioritization Using RSVP (RFC 2208) Congestion Control Algorithms Jitter Occurs when packets arrive out of order as each one may take different route Buffering helps to solve the Jitter problem. Wait for the slowest packet to arrive and play in the correct sequence But Increasing Buffer size will increase Delay

Issues in VoIP………….contd. Packet Loss When the receiving side buffer is full some of the packets are discarded Packet loss is 5% is tolerable If greater than 5% conversation becomes choppy One Solution is to replay the last packet received Can send info about nth packet along with n+1th packet, but this will waste bandwidth Echo Some percentage of the signals are reflected back to the user. Very annoying Can be eliminated using Echo Cancellers

Issues in VoIP………….contd Bandwidth Optimizations Converge Voice and Data to save Bandwidth Main Cause of Delay, Congestion and packet loss is insufficient bandwidth High Bandwidth is required for good quality calls(64kbps) Various Compression Standards can be used to reduce required bandwidth G.711 Audio Codec,3.1 KHz at 48, 56, and 64 Kbps (normal telephony). G.722 Audio Codec, 7 KHz at 48, 56, and 64 Kbps G.728 Audio Codec, 3.1 KHz at 16 Kbps. G.723 Audio Codec, for 5.3 and 6.3 Kbps modes G.729 Audio Codec

Issues in VoIP……….contd Bandwidth Optimizations…. Silence Suppression is another technique which can significantly save bandwidth With Silence Suppression only active voice signals are sent over the network. Since voice conversation typically consists of 50 to 60% silence, the amount of bandwidth conserved is considerable. With silence suppression, two conversations can share the bandwidth normally used by one conversation. Interoperability and Scalability VoIP products are still proprietary Products of different vendors should be able to work together All the vendor should follow a particular standard VoIP should interoperate with the existing PSTN network The current technology should be scalable to future innovations

Basic VoIP Services Call Forwarding Call Waiting Do not Disturb Multimedia Conferencing Caller/Callee Authentication Voice mail Buddy List Call transfer

SIP: Session Initiation Protocol Light weight generic signaling protocol Product of IETF MMUSIC Working group RFC 2543 Main functions Initiate and terminate multimedia sessions with users Locate users, allowing for mobility and proxying HTTP like -INVITE: initiation a session - BYE: terminate a session - REGISTER: register addresses with server

SIP: Session Initiation Protocol Supports unicast as well as multicast Port Number (UDP/TCP) 5060 Multicast address- 224.0.1.75 Invite to new and ongoing conferences Provides call control(hold, transfer, forward, media change..) It can web oriented..can leverage features of web Infrastructure:security, “CGI-BIN”, electronic payments, cookies etc Easily extends to presence information( buddy list) SIP can also be used for Internet real time fax delivery SIP provides also caller and callee authentication SIP supports personnel mobility: the ability to reach a called party under single, location independent address even when the user changes terminal

SIP: Session Initiation Protocol SIP is Text based (~HTTP) The various methods in SIP are INVITE : initiate call ACK confirm final response BYE terminate (or transfer) call CANCEL cancel “searches” or “ringing” OPTIONS features supported by the other side REGISTER register with location service SIP also support various IN services like distinctive ringing, #800, call center etc

H.323 H.323 is the set of standards defining multimedia communications and conferencing over packet-based network First accepted by ITU-T in 1996 H.323 has 4 important components Terminals Gateway Gatekeepers Multiple control Units(MCUs)

H.323 …contd Terminals: Terminals Provide Real-time communications. They must support voice/video/data communications. The most common H.323 Terminal today is Microsoft net meeting Gateway: H.323 gateways provide services to H.323 clients so that they can communicate with non H.323 clients. The gateway must provide translations between different transmission formats, communication procedures, and audio codecs Gatekeepers: They provide call control services for H.323 endpoints, such as address translation and bandwidth management. If they are present in the network the endpoint must use their services. The H.323 standards define mandatory as well as optional services for the gatekeeper. Multiple Control Units(MCUs): They provide conferences of 3 or more endpoints

Comparison of SIP and H.323 SIP H.323 IETF MMMUSIC ITU_T Proxy Gatekeeper Large and small conference MCU based control unit Firewall friendly complex, multiple protocols Multicast signaling ----- Personnel mobility Not(yet) SSL, HTTP security work in progress Post dial delay:1.5RTT 8.5RTT Text based binary based. 60+ pages 200 complex pages!!! Easy implementation , Debugging very complex Can be implemented using Java, Tcl,Perl mostly only in C/C++

Supporting VoIP Protocols SDP(Session Description Protocol) Intended for describing multimedia sessions RSVP(Resource Reservation Protocol) for prioritization of voice packets RTP(Real time Protocol) for sequencing of packets H.225 : Specifies messages for call control including signaling, registration and admission H.245 -Specifies messages for opening and closing channels H.261 -Video codec for audiovisual services at 64kbps H.263 - New Video codec for video over POTS

Current Research and Development Long term benefits accrue from multimedia and multi-service applications. Voice/Video is an integral part of conferencing systems that may include shared screens, etc., IP phones are under development. It’s a device that transports voice over a data network using IP protocol. More work needed on voice quality. It doesn’t match PSTN currently. Try to combine WWW access to information with a voice call button that allows immediate access to a call center agent from PC. Try an evolve on an integrated infrastructure that supports all forms of communication and allow more standardization.

Security in VoIP VoIP protocols provide hop-by-hop encryption and authentication Proxy Authentication in case of SIP End to end cryptography ..PGP etc Caller and Callee Authentication

A new Product Idea.. A hotmail type service video/voice has been envisioned Supporting voice/video chats Voice/video mails Targeting advertising using voice mail servers Premium services Web based Supported by any generic browser

Alternatives to Voice over IP There are alternatives which are simpler to implement today, and lack QoS issues which make VoIP difficult today Voice over Frame Relay Voice over ATM Reasonably mature technology New standards are evolving Best solution is to build hybrid ATM / FR networks easily to increase flexibility and manage QoS quite well.

Voice over ATM Accommodate mixed voice and data environment ATM has a number of classes of service - can give voice traffic higher priority ATM uses short, fixed length packets - enforces QoS based class, not on length of packet But is very expensive Will it be an alternative for VoIP.?????????

Driving Factors of VoIP Growth of the Internet and wide usage of IP It is cheaper to make IP telephony call because there is no interconnect charge A circuit switched call takes up 64kbps whereas a VoIP uses only 6-8kbps Offer exciting new value adds in long term: IP multicast conferencing, distance learning applications, phone directories and screen popping via IP, voice web browsing(where the caller can interact with the web page by speaking commands) Finally it gives carriers ability to manage a single network handling both voice/video/data.

Thoughts and Ideas( how to improve) Cut unnecessary broadcasting packets – increase the capacity. Engineer the network. Packet should go through minimum hops. Increase the capacity of the congested link- avoid congestion. Use buffer to solve jitter. Go for an optimal buffer size

Standards groups on VoIP technology ITU-T - teleconferencing systems, protocols, and audio/video encoding International Multimedia Teleconferencing Consortium ( IMTC ) VoIP Forum - H.323 interoperability agreements European Telecommunications Standards Institute ( ETSI ) Internet Engineering Task Force ( IETF) – SIP, RTP, RSVP, ISSLL, etc. IEEE 802 (in particular, 802.1p/Q, and 802.3x)

Projected rise in market 34% of telephone calls will be carried via IP or packet networks by 2005 ( Probe Research) Convergence of data and voice is happening Major telecommunications companies (e.g.., AT&T, MCI) are focusing on VON in their strategic plans No longer voice traffic – its data traffic with data applications

Internet Telephony Market Growth

Future Challenges!!! Scalability and Performance : - High performance architecture that can scale to billions of ports Quality of Service (QoS ) : - Reliable data networking - Provide PSTN quality (voice and latency) Interoperability - Needs open standards and interfaces - Natural transition and migration Carrier grade support - Billing/ rating solutions, Network management

Conclusions VoIP is a new technology that enables voice/fax to be delivered using IP. The driving force for the technology is to avoid long distance charges in PSTN. VoIP is becoming a hot topic in relation to voice/data integration. Because of the rapid growth of data network it is possible to have voice over IP An alternative to Public Switched Telephone Network. It offers exciting new added values like – telephony distance learning, phone directories, “ voice web browsing”, multicast conferencing. Internet-aware telephones are on the horizon. VoIP is a growing technology and very practical for large companies and organizations.

References: “ VoIP – Technical and Application Overview.” Queesnland university of Technology “ VoIP – Senior Design Project.” Stevens institute of Technology “ Internet Telephony Overview.” Intelliswitch.com/it.html Intelli-switch 7 Dec. 1998 http://www.cs.uml.edu/~akapadia/second_paper/index.html “ VoIP- Sizzle or Steak.” http://members.aol.com/RoyM11/LoopCo/VOIP.html “ Comparison of H.323 and SIP for IP Telephony Signaling.” Ismail Dalgic, Hanlin Fang http://www.cs.columbia.edu/~hgs/sip/ Impact Of H.323 On VON http://www.micom.com/international/tech/von_h323.htm SECOND PAPER ON VOICE OVER IP http://www.cs.uml.edu/~akapadia/second_paper/index.html VoIP - Technical and Application Review http://sky.fit.qut.edu.au/~rolf/itn540/gallery/a199/anoussa/itn540~1.htm Voice Over IP Network Design http://www.zetacom.demon.co.uk/infolibvipnd.htm Voice over IP Technologies: Ready For The Enterprise? http://www.hermesgroup.com/whitepapers/VoIP/voip.htm Voice Over IP Telephony: Sizzle or Steak? http://members.aol.com/RoyM11/LoopCo/VOIP.html http://www.anite.com.au/Events/VoIP.htm http://www.pcs.ellemtel.net/theo/Talks/EBI/sld001.htm

Voice over IP: Issues and Protocols

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