Telecommunications Blog

Real-Time Transport Protocol (RTP)

RTP is used for transmitting audio and video packets between communicating computers. RTP is defined in RFC 3550 and "profiles", which specify how to packetize various audio and video codecs used in an RTP session, are defined in RFC 3551.

 RTP also addresses issues like packet order and provides mechanisms to help address delay and jitter. RTP was improved upon with the result being called Secure RTP which provides for encryption, authentication, and integrity of the audio and video packets transmitted between communicating devices.

H.323 

Developed by Study Group 16 of the ITU-T, H.323 is an umbrella standard encompassing
many subcomponent standards and annexes for transmitting multimedia (voice, video and data) across packet based networks. The two main subcomponent protocols are H.225 (call control), & H.245 (bearer control and capabilities exchange). H.225 consists of two main parts Q.931 (basic call control as used in ISDN networks) and RAS (Registration, Admission & Status).

H.323 also references other standards such as Real Time Protocol (RTP), G.xxx audio codecs, H.26x video codecs and T.120 real time data conferencing protocol.

SIP 

Developed by IETF, SIP is a mechanism to initiate, terminate & modify sessions in an IP
network. It uses a client / server architecture and the protocol is request-response based. It
enables personal mobility by tracking down users and delivering calls to an endpoint. It is a
lightweight, text-based protocol and reuses much of the construct of other internet protocols
such as HTTP and SMTP. SIP does not know about the underlying details of a session and relies on IETF protocol Session Description Protocol (SDP) to describe the session. It also interworks with other IETF protocols such as Megacop, RTP, RTSP, RSVP and SAP.


Both H.323 and SIP can be referred to as "intelligent endpoint protocols". What this means is that all of the intelligence required to locate the remote endpoint and to establish media streams between the local and remote device is an integral part of the protocol.

MGCP & Megacop/H.248 

There is another class of protocols which is complementary to H.323 and SIP referred to as "device control protocols". Those protocols are H.248 and MGCP. 

MGCP (Media Gateway Control Protocol) & Megacop/H.248 are relatively low level, master/slave protocols used between Call Agents and MGs. MGCP is not officially a standard but does exist as an informational RFC (RFC 3435). Megacop (IETF RFC 3015) & H.248 (ITU Rec.) are the same protocol developed by an IETF and ITU collaboration. It is derived from, and draws heavily from MGCP but adds several new enhancements.

Some service providers provide users with devices that implement H.248 or MGCP (or comparable protocols). In the core of the network, some device serving as the MGC provides the H.323 or SIP logic necessary to properly terminate VoIP calls around the world.

VoIP OSI Protocol Stack

Telecommunications Blog

I will explain QOS at next article .

What is VoIP

Around 1995 there was a revolutionary advancement in the way traditional voice traffic was
carried when voice, video and data were transmitted over a single ‘converged’ network. The
terms Voice Over IP (VoIP) and IP Telephony (IPT) were introduced to describe this method of
transport of voice signals as well as fax, DTMF tones, signalling traffic, and network management
traffic.


There are a number of protocols that may be employed in order to provide for VoIP communication services like :

• SIP (Session Initiation Protocol) / RTP  (Real-Time Protocol)
• IMS
• H.323
• H.248 / MGCP   ( comparable protocols )
• Skype

VoIP can deliver voice, video, and data communications capabilities across a number of separate devices that work together, such as a mobile phone, a PC, and even an HD TV!.

You can enjoy VoIP sevice by using:

1. An analog Telephone Adapter (ATA). It connected existing telephone jack, so you no need to change your old telephone hardware (system).
2. Dedicated VoIP Phones. It is a Phones that create specially for IP/Packet Switch connection. Just plug this phone to your IP/internet network then you can enjoy VoIP service.
3. A Softphone. It is a piece of software that installed at PC/Laptop to make you can enjoy VoIP service without dedicated hardware.

I will explain the protocols at the next article.

Its full name is 3GPP Long Term Evolution for the Universal Mobile Telecommunications System, or 3GPP UMTS LTE for short. LTE (Long Term Evolution) is a wireless broadband technology designed to support roaming Internet access via cell phones and handheld devices.


Where the current generation of mobile telecommunication networks are collectively known as 3G, LTE is marketed as 4G.


Requirements :


LTE Specifications :



 LTE uses two different types of air interfaces (radio links), one for downlink (from tower to device), and one for uplink (from device to tower).For the downlink, LTE uses an OFDMA (orthogonal frequency division multiple access) air interface For the uplink (from device to tower), LTE uses the DFTS-OFDMA (discrete Fourier transform spread orthogonal frequency division multiple access) scheme of generating a SC-FDMA (single carrier frequency division multiple access) signal. 

The major difference between the OFDMA signal for downlink and the SC-FDMA signal for uplink is that it uses a discrete Fourier transform function on the data to convert it into a form that can be used to transmit.