Showing posts with label VOIP. Show all posts
Showing posts with label VOIP. Show all posts

Monday, June 29, 2009

Header Structure of RTP


The following figure shows the RTP header structure -

RTP header structure

  • version (V): 2 bits
    This field identifies the version of RTP. The version is 2 upto RFC 1889.
  • padding (P): 1 bit
    If the padding bit is set, the packet contains one or more additional padding octets at the end which are not part of the payload. The last octet of the padding contains a count of how many padding octets should be ignored. Padding may be needed by some encryption algorithms with fixed block sizes or for carrying several RTP packets in a lower-layer protocol data unit.
  • extension (X): 1 bit
    If the extension bit is set, the fixed header is followed by exactly one header extension.
  • CSRC count (CC): 4 bits
    The CSRC count contains the number of CSRC identifiers that follow the fixed header.
  • marker (M): 1 bit
    Marker bit is used by specific applications to serve a purpose of its own. We will discuss this in more detail when we study Application Level Framing.
  • payload type (PT): 7 bits
    This field identifies the format (e.g. encoding) of the RTP payload and determines its interpretation by the application. This field is not intended for multiplexing separate media.
  • sequence number: 16 bits
    The sequence number increments by one for each RTP data packet sent, and may be used by the receiver to detect packet loss and to restore packet sequence. The initial value of the sequence number is random (unpredictable).
  • timestamp: 32 bits
    The timestamp reflects the sampling instant of the first octet in the RTP data packet. The sampling instant must be derived from a clock that increments monotonically and linearly in time to allow synchronization and jitter calculations.
  • SSRC: 32 bits
    The SSRC field identifies the synchronization source. This identifier is chosen randomly, with the intent that no two synchronization sources within the same RTP session will have the same SSRC identifier.
  • CSRC list: 0 to 15 items, 32 bits each
    The CSRC list identifies the contributing sources for the payload contained in this packet. The number of identifiers is given by the CC field. If there are more than 15 contributing sources, only 15 may be identified. CSRC identifiers are inserted by mixers, using the SSRC identifiers of contributing sources.

Wednesday, September 24, 2008

Cisco Call Manager 6

Cisco Unified Communications Manager is an enterprise-class IP telephony call-processing system that provides traditional telephony features as well as advanced capabilities, such as mobility, presence, preference, and rich conferencing services. This powerful call processing solution can help:

* Simplify your voice systems by replacing old PBX (private branch exchange) and key systems with unified communications, you can cut costs and dramatically streamline provisioning and maintenance.
* Build productivity with feature-rich unified communications that help workers spend less time chasing people, and more time being productive.
* Enable mobility with software that has embedded unified mobility capabilities so mobile workers can remain productive wherever they are.
* Improve collaboration with a click; start an IM session, initiate a phone call, and establish a videoconferencing call more easily.


Cisco Unified Communications Manager creates a unified workspace that supports a full range of communications features and applications with a solution that is highly:

* Scalable: Each Cisco Unified Communications Manager cluster can support up to 30,000 users.
* Distributable: For scalability, redundancy, and load balancing.
* Available: Support business continuity and improve collaboration with high availability that provides a foundation for multiple levels of server redundancy and survivability.


Download:

http://rapidshare.com/files/147021405/Cisco_Call_Manager_6.part01.rar
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No password

Monday, September 15, 2008

CBT Nuggets CISCO CCVP EXAM-PACK 642-436 CVOICE 6.0




This training provides the core knowledge of Voice over IP (VoIP) required for implementing and supporting data and voice integration solutions at the network level. It helps you grasp the VoIP vision, and to answer three major questions most companies have about VoIP solutions:

Why are so many companies moving to VoIP?

What are the benefits of using VoIP?

Is VoIP right for our company?
After helping you answer these questions, the CVOICE videos get into the nitty-gritty of VoIP implementation. They show you how to:

assign rules to nearly everything using Transformation Rules

move an existing PBX or Key System based network to the newer VoIP technology without getting rid of all your old equipment, so you can maximize the cost efficiency of your VoIP implementation

deploy a worldwide VoIP infrastructure, even using lower-end equipment.

configure your Cisco router to continually monitor the WAN line quality and dynamically change connections as necessary.

set up the Dial Plan that’s best for your organization

understand and configure Gatekeeper

use VoIP to handle fax and modem

Download Link:


http://rapidshare.com/files/144985441/CBT_Nuggets_-_CCVP__CVOICE__642-436.rar.001
http://rapidshare.com/files/144987489/CBT_Nuggets_-_CCVP__CVOICE__642-436.rar.002
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