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VoIP Basics

Rajib DekaSr. ProgrammerSiemens Ltd. Chennai-100.

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Agenda Internet Basics Protocol Layering Voice Over IP VoIP Architecture VoIP Network VoIP Protocols SIP Basics Conclusion

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Internet Basics The basic building block of networks is the IP datagram.

Analogy to datagram - a postcard with Destination address Return address A small amount of text

A postcard might inform you of a friend’s holiday travels or remind you of a dentist’s appointment.

The postal service doesn’t care which application (friend or dentist) sent the postcard—it just carries processed wood pulp with black marks.

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Protocol Layering

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Voice Over IP - Introduction A recent application of Internet technology – Voice

over IP (VoIP): Transmission of voice over Internet.

How VoIP works Continuously sample audio Convert each sample to digital form Send digitized stream across Internet in packets Convert the stream back to analog for playback

Why VoIP IP telephony is economic; High costs for traditional

telephone switching equipments. Call setup: call establishment, call termination, etc. Backward compatibility with existing PSTN (Public Switched

Telephone Network)

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Voice Over IP - Introduction IP Telephony Standards:

ITU (International Telecommunication Union) controls telephony standards.

IETF (Internet Engineering Task Force) controls TCP/IP standards.

Audio is encoded using a well-known standard such as Pulse Code Modulation (PCM).

UDP is used for transport: lower overhead: audio must be played as it arrives. Playback cannot be stopped to wait for a retransmitted

packet.

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VoIP Architecture VoIP Server:

Media Gateway. Media Gateway Controller. Signaling Gateway. IP PBX and Proxy.

VoIP Client: Soft Phones. IP Phones.

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VoIP Network

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VoIP Network

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VoIP Protocols Main complexity of VoIP: Call setup and call management.

The process of establishing and terminating a call is called Signaling.

In traditional telephone system, signaling protocol is SS7.

In VoIP, signaling protocols are: SIP (Session Initiation Protocol), by IETF H.323, by ITU Megaco & MGCP, jointly by IETF and IUT.

Audio Signaling: RTP: Real-time Transport Protocol. RTCP: Real Time Control Protocol.

VoIP signaling protocols should be able to interact with SS7.

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SIP Basics SIP: Session Initiation Protocol. Invented by IETF. SIP defines three main elements that comprise a

signaling system: User Agent: IP phone or applications Location servers: stores information about user’s location

or IP address Support servers:

Proxy Server: forwards requests from user agents to another location.

Redirect Server: provides an alternate called party’s location for the user agent to contact.

Registrar Server: receives user’s registration requests and updates the database that location server consults.

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SIP Characteristics Operates at the application layer.

Encompasses all aspects of signaling, e.g. location of called party, ringing a phone, accepting a call, and terminating a call.

Provides services such as call forwarding.

Relies on multicast for conference calls.

Allows two sides to negotiate capabilities and choose the media and parameters to be used.

SIP URI is similar to email address. (with prefix “sip:”) E.g. sip:rajib.deka@siemens.com

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SIP Methods Six basic message types, known as methods:

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SIP Session User agent A contacts DNS

server to map domain name in SIP request to IP address.

User agent A sends a INVITE message to proxy server that uses location server to find the location of user agent B.

Call is established between A and B. Then media session begins.

Finally, B terminates the call by sending a BYE request.

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RTP and RTCP RTP used to send real-time streams of data across a network is

simply called the Real Time Protocol (RTP for short). RTP has been originally defined by IETF.

RTCP accompanies RTP and is used to transmit control

information about the RTP session. RTCP packets are send only from time to time since there is a recommendation that the RTCP traffic should consume less than 5 percent of the session bandwidth. The most important content types carried in RTCP packets include:

Information about call participants (for example, name and e-mail address)

Statistics about the quality of the transmission (for example inter-arrival jitter and the number of lost packets).

RTCP to monitor Quality of Service (QoS).

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VoIP and QoS QoS (Quality of Service) is a major issue in VOIP

implementations. The issue is how to guarantee that packet traffic for a voice or other media connection will not be delayed or dropped due interference from other lower priority traffic.

Things to consider are Latency: Delay for packet delivery Jitter: Variations in delay of packet delivery Packet loss: Too much traffic in the network causes the

network to drop packets Burstiness of Loss and Jitter: Loss and Discards (due to

jitter) tend to occur in bursts.

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VoIP Developer’s choice Language:

C, C++, Java for Core protocol stack development. Java, C# for middle tier or application development.

Open Source VoIP Servers (Linux Based) Asterisk PBX (Multi protocol support) OpenSIPS (for SIP only) sipXecs (for SIP only)

VoIP Clients X-lite. VoIP Communicator.

Open Source SIP stack JainSIP sipXecs PJSIP

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Conclusion IP telephony or VoIP refers to the transmission of voice

telephone calls over IP networks.

Hot area both in research and market because of low cost

Challenge in backward compatibility with PSTN

The complexity of IP telephony is on signaling. Both ITU and IETF propose signaling standards.

H.323, by IUT SIP, by IETF, offering similar functions to H.323, but simpler than

H.323. Both are competing to be recognized as #1 signaling protocol.

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