VOIP INTRODUCTION
Huy Nguyen
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Outline
l VoIP Overview
l SIP Overview
l RTP Overview
l SIP Message
l SIP Call Feature
l SIPTrunking
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VoIP Overview
• VoIP fundamental
• VoIP Network Architecture
• VoIP Protocols
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VoIP Overview
l Multi-service via a single access line
l Any transport media
l Capture new markets
l Multi-functional controller
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VoIP Overview
SIP Proxy/Redirect server
IP Clients and Switch
Softswitch
Terminals
PSTN
Registration/Routing
Server
Switch
H.323 / SIP
H.323 / SIP
IP PC Client
H.323 / SIP
S
IE
N
IX
MENS
DORF
H.323 / SIP
H.323 /
SIP
IP PBX
IP Phones
S
IIX
N MO
E
DER
NF
S
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VoIP network Architecture
Management
Switch Control
Core
Switch Switch
Access
S EMENS
NXDORF
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VOIP DIAGRAM
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VOIP PROTOCOLS VIEW
Call Control and Signaling Signaling and Media
Gateway Control
Audio/
H.323 Video
H.225
H.245 Q.931 RAS SIP MGCP RTP RTCP RTSP
TCP UDP
IP
H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.
H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.
SIP supports TCP and UDP.
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SIP overview
l SIP Architech
l H323 vs SIP
l How SIP message works?
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Why SIP?
Χ SIP is emerging as the de facto
standard for a Multi-media
services delivery architecture
Χ SIP lever ages existing internet
tools and protocols (DNS, SDP,
MIME types...)
Χ SIP is the right Protocol for the
MSF proposed NGN distributed
architecture
Χ Decentralized introduction of
advanced applications
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Fig. 1 (SN2050EU01SN_0010 VoBB and NextGen Applications SIP Based, 5)
SIP Overview
l Part of IETF conference control architecture:
l SIP for advertising multimedia sessions
l RTSP for controlling delivery of streaming media [RFC 2326]
l SDP for describing multimedia sessions [RFC 2327]
l RTP for transporting real-time data and providing QoS
feedback [RFC 1889]
l HTTP-like [RFC 2616]
l URI and URL [RFC 2396]
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SIP vs H.323
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SIP vs H323
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RTP Overview
l Delivering audio and video over IP networks.
l Streaming media, such as telephony, video teleconference
applications, television services and web-based push-to-talk
features
l Conjunction with the RTP Control Protocol (RTCP).
l Was developed by the Audio-Video Transport Working Group
of the Internet Engineering Task Force (IETF)
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Protocol Components
l The data transfer protocol: include timestamps , sequence
numbers and the payload format .
l A control protocol : quality of service (QoS) feedback and
synchronization between the media streams
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SIP Components
l User Agent
l Proxy server
l Registrar
l Redirect server
l Session border controllers (SBC)
l Gateways
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SIP Message
l REGISTER: Used by a UA to indicate its current IP address and
the URLs for which it would like to receive calls
l INVITE: Used to establish a media session between user agents.
l ACK: Confirms reliable message exchanges.
l CANCEL: Terminates a pending request.
l BYE: Terminates a session between two users in a conference.
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SIP Message
l OPTIONS: Requests information about the capabilities of a
caller, without setting up a call.
l PRACK (Provisional Response Acknowledgement): PRACK
improves network reliability by adding an acknowledgement
system to the provisional Responses (1xx). PRACK is sent in
response to provisional response (1xx
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SIP Responses
l Provisional (1xx): Request received and being processed.
l Success (2xx): The action was successfully received,
understood, and accepted.
l Redirection (3xx): Further action needs to be taken (typically
by sender) to complete the request
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SIP Responses
l Client Error (4xx): The request contains bad syntax or cannot
be fulfilled at the server.
l Server Error (5xx): The server failed to fulfill an apparently
valid request.
l Global Failure (6xx): The request cannot be fulfilled at any
server.
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SIP MESSAGES
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SIP Call Feature
l SIP basic.
l SIP via Proxy.
l SIP via gateway.
l SIP in redirect.
l SIP register.
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SIP Call Feature
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SIP Call Feature
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SIP Call Feature
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SIP Call Feature
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SIP TRUNKING
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SIP Trunking Overview
l Trunk : VoIP Trunk is analogous to a phone line in a PSTN
network
l Multi-channels
l Via IP network or landline(analog trunk)
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Reference
http://toncar.cz/Tutorials/VoIP/
VoIP_Protocols_SIP_Call_Flow.html
http://en.wikipedia.org/wiki/Voice_over_IP
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