2. Agenda
Internet Basics
Protocol Layering
Voice Over IP
VoIP Architecture
VoIP Network
VoIP Protocols
SIP Basics
Conclusion
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3. 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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5. 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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6. 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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7. 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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10. 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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11. 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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12. 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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13. SIP Methods
Six basic message types, known as methods:
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14. 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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15. 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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16. 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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17. 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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18. 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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