ITNW 1380 COOPERATIVE EDUCATION –ITNW 1380 COOPERATIVE EDUCATION –

NETWORKING NETWORKING

Spring 2009Spring 2009SeminarSeminar # 4 # 4

VOIP Network SolutionsVOIP Network Solutions

Unified Communications and Collaboration Solution

Unified Communications IP Convergence: Voice, data & video on the same

network. Voice: Voice over IP - VOIP.

Collaboration Solution IM: Yahoo, MSN, Novell, Skype... 3G Wireless: WCDMA, UMTS. 4G Wireless: Wireless LAN, WIFI... Blackbery, Motorola, Iphone Web & Email

Today's Collaboration Solution

VOIP

VOIP

VOIP Definition VOIP Service Providers VOIP Benefits VOIP Protocols VOIP Networks VOIP Open Source Software

VoIP Definition

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

VOIP Definition VOIP Service Providers VOIP Benefits VOIP Protocols VOIP Networks VOIP Open Source Software

VoIP Service providers

With VoIP you can call PC to PC (Softfone), PC to IP phone, or IP phone to ordinary phone over Internet.

VoIP is popular international calling. VoIP enables you to call from virtually anywhere. VoIP provider offer low rates or free service deals – Free

World Dialup & Skype, Yahoo Voice...

VOIP

VOIP Definition VOIP Service Providers VOIP Benefits VOIP Protocols VOIP Networks VOIP Open Source Software

VOIP Benefits

Lower long distance rates: requires only Internet connection.

Simplicity: VoIP transmission combines voice & data. Capacity: VoIP better uses you network for less. Global outsourcing: International call centers rely on

VoIP. Automatic routing: Receive calls automatically to your

VoIP phone. Portability: Travel with your VoIP phones.

VOIP

VOIP Definition VOIP Service Providers VOIP Benefits VOIP Protocols VOIP Networks VOIP Open Source Software

VoIP Protocols

H.323 Session Initiation Protocol (SIP)

H.323 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.)

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:

Video codec: H.261, H.263, H.264 Audio codec: G.711, G.729 (including G.729a),

G.723.1, G.726 Text codecs: T.140

H.323 (Cont.)

H.323 Architecture:

The H.323 system defines several network elements that work together in order to deliver rich multimedia communication capabilities

Terminals, Multi-point Control Units (MCUs), Gateways, Gatekeepers, and Border Elements.

A complete, sophisticated H.323 protocol stack

VoIP Protocols (Cont.)

H.323 Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP)

Purpose of SIP SIP URI SIP Network Elements SIP Messages Transactions Dialogs Typical SIP Scenarios

Purpose of SIP

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.)

Two protocols that are most often used along with SIP are RTP and SDP. 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.

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.)

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

SIP entities are identified using SIP URI (Uniform Resource Identifier).

A SIP URI has form of sip:username@domain, for instance, sip:thanhnc@saigontech.edu.vn.

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

Basic SIP elements are:

User Agents: 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: 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.)

Registrar:

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.

Purpose of the location database is to map sip:thanhnc@saigontech.edu.vn to sip:thanhnc@1.2.3.4:5060.

SIP Messages

SIP Requests:

ACK This message acknowledges receipt of a final response to INVITE.

BYE Bye messages are used to tear down multimedia sessions.

CANCEL Cancel is used to cancel not yet fully established session.

REGISTER Purpose of REGISTER request is to let registrar know of current user's location.

Transactions

SIP is transactional protocol. A transaction is a sequence of SIP messages exchanged

between SIP network elements. SIP messages:

SIP user#1 SIP user#2 INVITE -----------------------------> <----------------------------- 200 OK ACK ------------------------------>

SIP Messages

Dialogs

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

Registration Session Invitation Session Termination Record Routing Event Subscription and Notification Instant Messages

VOIP

VOIP Definition VOIP Service Providers VOIP Benefits VOIP Protocols VOIP Networks VOIP Open Source Software

VoIP Network

Solution for Enterprise from AudioCodes Ltd.:

IP-PBX IP Telephony Survivable Network PBX VoIP Networking Contact Center for Enterprises Unified Messaging for Enterprises Conferencing for Enterprises

IP-PBX

IP Telephony Survivable Network

PBX VoIP Networking

Contact Center for Enterprises

Unified Messaging for Enterprises

Conferencing for Enterprises

VOIP

VOIP Definition VOIP Service Providers VOIP Benefits VOIP Protocols VOIP Networks VOIP Open Source Software

OpenSER:

SER – SIP Express Router (Registrar/Proxy/Redirect server)

Support database backends: MySQL, Oracle, Postgres.

RTP Proxy, NAT traversal Interoperability with Cisco, Microsoft. PingTel,

Siemens, Xten and many others. http://www.iptel.org/ser

VOIP Open Source Software

VOIP Open Source Software (Cont.)

Asterisk PBX:

Most popular open source VOIP software. IP PBX which connect users over IP to PSTN, T1/E1. Media gateway, bridge the legacy PSTN to IP

telephony. Media server with IVR, voice mail, automated

attendant, unified messaging. Call Center, ACD, advance skills-based routing. http://www/asterisk.org

References

SIP: http://www.sip.org/ SER: http://www.iptel.org/ser AudiCodes:http://www.audiocodes.com/solutions Asterisk: http://www.asterisk.org

Questions