Transcript sip:username@domain

ITNW 1380 COOPERATIVE EDUCATION –
NETWORKING
Spring 2009
Seminar # 4
VOIP Network Solutions
Unified Communications and Collaboration
Solution
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Unified Communications
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IP Convergence: Voice, data & video on the same
network.
Voice: Voice over IP - VOIP.
Collaboration Solution
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IM: Yahoo, MSN, Novell, Skype...
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3G Wireless: WCDMA, UMTS.
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4G Wireless: Wireless LAN, WIFI...
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Blackbery, Motorola, Iphone Web & Email
Today's Collaboration Solution
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VOIP
VOIP
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VOIP Definition
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VOIP Service Providers
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VOIP Benefits
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VOIP Protocols
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VOIP Networks
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VOIP Open Source Software
VoIP Definition
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VoIP (Voice Over Internet Protocol).
Voice transmission over packet based network such as
Internet, corporate intranet, LAN, WAN.
VoIP known as Internet telephony.
Integrate VoIP enabled voice signals with faxes & data
into a unified network.
Telephone conversation over Internet (IP Telephony).
VOIP
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VOIP Definition
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VOIP Service Providers
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VOIP Benefits
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VOIP Protocols
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VOIP Networks
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VOIP Open Source Software
VoIP Service providers
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With VoIP you can call PC to PC (Softfone), PC to IP
phone, or IP phone to ordinary phone over Internet.
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VoIP is popular international calling.
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VoIP enables you to call from virtually anywhere.
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VoIP provider offer low rates or free service deals – Free
World Dialup & Skype, Yahoo Voice...
VOIP
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VOIP Definition
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VOIP Service Providers
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VOIP Benefits
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VOIP Protocols
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VOIP Networks
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VOIP Open Source Software
VOIP Benefits
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Lower long distance rates: requires only Internet
connection.
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Simplicity: VoIP transmission combines voice & data.
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Capacity: VoIP better uses you network for less.
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Global outsourcing: International call centers rely on
VoIP.
Automatic routing: Receive calls automatically to your
VoIP phone.
Portability: Travel with your VoIP phones.
VOIP
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VOIP Definition
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VOIP Service Providers
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VOIP Benefits
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VOIP Protocols
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VOIP Networks
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VOIP Open Source Software
VoIP Protocols
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H.323
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Session Initiation Protocol (SIP)
H.323
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Published by Telecommunication Standardization Sector
(ITU-T) in November 1996.
ITU-T defines H.323 protocols to provide audio-visual
communication sessions on any packet network.
Widely implemented by voice and videoconferencing
equipment manufacturers.
Widely deployed worldwide by service providers and
enterprises for both voice and video services over
Internet Protocol (IP) networks.
H.323 (Cont.)
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Suited for transmitting calls across networks using a
mixture of IP, PSTN, ISDN, and QSIG over ISDN.
Within the context of H.323, an IP-based PBX might be
an H.323 Gatekeeper.
Codecs:
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Video codec: H.261, H.263, H.264
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Audio codec: G.711, G.729 (including G.729a),
G.723.1, G.726
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Text codecs: T.140
H.323 (Cont.)
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H.323 Architecture:
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The H.323 system defines several network elements
that work together in order to deliver rich multimedia
communication capabilities
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Terminals, Multi-point Control Units (MCUs),
Gateways, Gatekeepers, and Border Elements.
A complete, sophisticated H.323 protocol stack
VoIP Protocols (Cont.)
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H.323
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Session Initiation Protocol (SIP)
Session Initiation Protocol (SIP)
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Purpose of SIP
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SIP URI
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SIP Network Elements
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SIP Messages
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Transactions
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Dialogs
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Typical SIP Scenarios
Purpose of SIP
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Application-layer control protocol which has been
developed and designed within the IETF (Internet
Engineering Task Force).
The most important one is RFC3261 which contains the
core protocol specification.
The protocol is used for creating, modifying, and
terminating sessions with one or more participants.
Purpose of SIP (Cont.)
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Two protocols that are most often used along with SIP
are RTP and SDP.
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RTP – Real Time Protocol is used to carry the real-time
multimedia data (including audio, video, and text), the
protocol makes it possible to encode and split the data into
packets and transport such packets over the Internet.
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SDP – Session Description Protocol, which is used to
describe and encode capabilities of session participants
(negotiation of codecs used to encode media so all
participants will be able to decode it).
Purpose of SIP (Cont.)
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SIP is based on HTTP protocol.
SIP is used to carry the description of session parameters,
the description is encoded into a document using SDP.
SIP URI
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SIP entities are identified using SIP URI (Uniform
Resource Identifier).
A SIP URI has form of sip:username@domain, for
instance,
sip:[email protected].
As we can see, SIP URIs are similar to e-mail addresses,
it is, for instance, possible to use the same URI for e-mail
and SIP communication, such URIs are easy to
remember.
SIP Network Elements
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Basic SIP elements are:
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User Agents:
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Internet end points that use SIP to find each other
and to negotiate a session characteristics are called
user agents, IPphones,PSTN gateways, PDAs)
Proxy Servers:
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User agents can send messages to a proxy server.
They perform routing of a session invitations
according to invitee's current location,
authentication, accounting and many other
important functions
SIP Network Elements (Cont.)
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Registrar:
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The registrar is a special SIP entity that receives
registrations from users, extracts information about
their current location (IP address, port and username
in this case) and stores the information into location
database.
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Purpose of the location database is to map
sip:[email protected] to
sip:[email protected]:5060.
SIP Messages
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SIP Requests:
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ACK This message acknowledges receipt of a final
response to INVITE.
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BYE Bye messages are used to tear down multimedia
sessions.
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CANCEL Cancel is used to cancel not yet fully
established session.
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REGISTER Purpose of REGISTER request is to let
registrar know of current user's location.
Transactions
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SIP is transactional protocol.
A transaction is a sequence of SIP messages exchanged
between SIP network elements.
SIP messages:
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SIP user#1
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INVITE ----------------------------->
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SIP user#2
<----------------------------- 200 OK
ACK
------------------------------>
SIP Messages
Dialogs
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A dialog represents a peer-to-peer SIP relationship
between two user agents.
For instance, INVITE message establishes a dialog,
because it will be later followed by BYE request which
will tear down the session established by the INVITE.
This BYE is sent within the dialog established by the
INVITE.
SIP Dialogs
Typical SIP Scenarios
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Registration
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Session Invitation
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Session Termination
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Record Routing
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Event Subscription and Notification
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Instant Messages
VOIP
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VOIP Definition
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VOIP Service Providers
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VOIP Benefits
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VOIP Protocols
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VOIP Networks
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VOIP Open Source Software
VoIP Network
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Solution for Enterprise from AudioCodes Ltd.:
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IP-PBX
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IP Telephony Survivable Network
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PBX VoIP Networking
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Contact Center for Enterprises
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Unified Messaging for Enterprises
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Conferencing for Enterprises
IP-PBX
IP Telephony Survivable Network
PBX VoIP Networking
Contact Center for Enterprises
Unified Messaging for Enterprises
Conferencing for Enterprises
VOIP
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VOIP Definition
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VOIP Service Providers
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VOIP Benefits
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VOIP Protocols
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VOIP Networks
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VOIP Open Source Software
VOIP Open Source Software
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OpenSER:
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SER – SIP Express Router (Registrar/Proxy/Redirect
server)
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Support database backends: MySQL, Oracle,
Postgres.
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RTP Proxy, NAT traversal
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Interoperability with Cisco, Microsoft. PingTel,
Siemens, Xten and many others.
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http://www.iptel.org/ser
VOIP Open Source Software (Cont.)
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Asterisk PBX:
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Most popular open source VOIP software.
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IP PBX which connect users over IP to PSTN, T1/E1.
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Media gateway, bridge the legacy PSTN to IP
telephony.
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Media server with IVR, voice mail, automated
attendant, unified messaging.
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Call Center, ACD, advance skills-based routing.
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http://www/asterisk.org
References
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SIP: http://www.sip.org/
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SER: http://www.iptel.org/ser
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AudiCodes:http://www.audiocodes.com/solutions
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Asterisk: http://www.asterisk.org
Questions