BZU PAGES: Find Presentations, Reports, Student's Assignments and Daily Discussion; Bahauddin Zakariya University Multan Right Header

HOME BZU Mail Box Online Games Radio and TV Cricket All Albums
Go Back   BZU PAGES: Find Presentations, Reports, Student's Assignments and Daily Discussion; Bahauddin Zakariya University Multan > Institute of Computing > Bachelor of Science in Information Technology > BsIT 5th Semester > Broadband Networks

Broadband Networks Sir Taimoor


Reply
 
Thread Tools Search this Thread Rate Thread Display Modes
  #1  
Old 31-10-2009, 01:23 AM
.BZU.'s Avatar


 
Join Date: Sep 2007
Location: near Govt College of Science Multan Pakistan
Posts: 9,693
Contact Number: Removed
Program / Discipline: BSIT
Class Roll Number: 07-15
.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute.BZU. has a reputation beyond repute
lectures H.323 and SIP (Session Initiation Protocol)

H.323 is an ITU VOIP protocol. It was created at about the same time as SIP, but was more widely adopted and deployed earlier. Today, most of the world's VoIP traffic is carried over H.323 networks, with billions of minutes of traffic being carried every month.

H.323's strengths lie in its ability to serve in a variey of roles, including multimedia communication (voice, video, and data conferencing), as well as applications where interworking with the PSTN is vital. H.323 was designed from the outset with multimedia communications over IP networks in mind, making it the perfect solution for real-time multimedia communication over packet-based networks.


SIP
SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services.

SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP -the session description protocol.

  • SIP is a text-based protocol that uses UTF-8 encoding
  • SIP uses port 5060 both for UDP and TCP. SIP may use other transports

SIP offers all potentialities of the common Internet Telephony features like:
  • call or media transfer
  • call conference
  • call hold

Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability.

SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP)

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

  • Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
  • Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported � recognizing that not all the parties can support the same level of features. For example video may or may not be supported; as any form of MIME type is supported by SIP, there is plenty of scope for negotiation.
  • Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
  • Call feature changes: A user should be able to change the call characteristics during the course of the call. For example, a call may have been set up as 'voice-only', but in the course of the call, the users may need to enable a video function. A third party joining a call may require different features to be enabled in order to participate in the call
  • Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

__________________
(¯`v´¯)
`*.¸.*`

¸.*´¸.*´¨) ¸.*´¨)
(¸.*´ (¸.
Bzu Forum

Don't cry because it's over, smile because it happened
Reply With Quote
Reply

Tags
h323, initiation, protocol, session, sip


Currently Active Users Viewing This Thread: 1 (0 members and 1 guests)
 
Thread Tools Search this Thread
Search this Thread:

Advanced Search
Display Modes Rate This Thread
Rate This Thread:

Posting Rules
You may not post new threads
You may not post replies
You may not post attachments
You may not edit your posts

BB code is On
Smilies are On
[IMG] code is On
HTML code is Off
Trackbacks are On
Pingbacks are On
Refbacks are On


Similar Threads
Thread Thread Starter Forum Replies Last Post
Chapter # 18, Designing a Routing Protocol Deployment bonfire Telecommunication Network Design 0 15-04-2011 06:22 PM
Network Layer: Internet Protocol bonfire Data Communication 0 18-03-2011 06:36 PM
SS7 Protocol Stack bonfire Digital Telephony 0 10-12-2010 04:45 PM
Dynamic Host Configuration Protocol .BZU. System Admin 0 05-02-2010 12:39 AM
Cisco to introduce new messaging protocol BSIT07-01 Daily News And halat-e-hazra 0 23-05-2008 04:45 PM

Best view in Firefox
Almuslimeen.info | BZU Multan | Dedicated server hosting
Note: All trademarks and copyrights held by respective owners. We will take action against any copyright violation if it is proved to us.

All times are GMT +5. The time now is 03:12 AM.
Powered by vBulletin® Version 3.8.2
Copyright ©2000 - 2024, Jelsoft Enterprises Ltd.