Technical Posts Archives | Page 15 of 23

26 July 201826 July 2018

Understanding REGISTER method

SIP is a peer-to-peer protocol where the roles client – server and exchangeable depending on who starts a session. In reality most deployments foresee a process called registration (method: REGISTER) which allows a central server (registrar) to store the location of a SIP User-Agent.

A SIP Phone is a client to the central Unified Communication Platform (registrar) – and the UC platform is a client to the SIP Server of the operator (registrar).

Once the SIP Server gets to know the location of an SIP Client, it can deliver calls and other messages to it from other Clients connected to the same Server.

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19 July 201820 July 2018

Analyzing SIP and SIP calls

This blog article will be dedicated to SIP and SIP calls. It is meant to provide an introduction for UC Engineers to the main headers and guide through the debugging of the most popular call scenarios that will be described in the next articles.

As we have seen in SIP and XMPP standards in Unified Communications and Media transfer in Unified Communications – SDP protocol, SIP (Session Initiation Protocol) and SDP-RTP have become the de facto replacement mode for traditional analog and digital lines provided by operators all over the world.

SIP is also the most popular signaling mode to handle VoIP calls.

Being able to understand SIP sessions has become for a Unified Communication Engineer as important as in the past for Telephony Experts to read signaling traces in Q.931 / Q.921 / DS1 / QSIG generated by T1 / E1 PRI – BRI ISDN lines.

Before Starting the Analysis

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6 July 201820 June 2018

RTP, RTCP and Jitter Buffer

In this blog article we continue to analyze RTP and RTCP and we will see why Jitter Buffer is important and how it affects call quality.

As we saw in the previous article — SDP is not able to transfer media–this task is delegated to protocols such as RTP or RTSP.

RTCP (or Real Time Control Protocol) provides different levels of feedback about the ongoing RTP Stream.

The goal of RTCP is to provide information to the remote endpoint about the quality of service of the ongoing communication.

This is done by providing regular statistics about the amount of packets received, jitter, and packets lost (either via network or discarded by the jitter buffer). Continue reading “RTP, RTCP and Jitter Buffer”

29 June 201829 June 2018

Media transfer in Unified Communications – SDP protocol

Media is another vital component of a Unified Communication system. Once signaling is in place and working between two endpoints, information about media capabilities can be transferred, eventually allowing for streaming audio, making video calls, or exchanging other information.

In this blog article we will analyze what technologies are used to transfer information about available media between endpoints.

SDP

SDP (Session Description Protocol) is a format for describing streaming media initialization parameters standardized by IETF in 1998.

What follows is the Session Description fields usage and an example:

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21 June 201822 June 2018

Introducing BOSH and WebSocket Transport protocols

This time we will talk about transport protocols over the web, in particular, about BOSH and WebSocket.

Besides TCP (XMPP/SIP) and UDP (SIP only) transports, two other transports, BOSH and WebSocket, are available which are embedded inside existing TCP/HTTP stacks.

BOSH

Bidirectional-streams Over Synchronous HTTP (BOSH) allows real-time communication between a browser and a web server. The browser connects to the server and will keep the connection open as long as it has no data to send. When data is available, the server sends it over the open HTTP connection and closes the connection itself. This reduces the number of requests, as the browser is not continuously polling the server. The server retains a cache of events that the client missed between reconnections.

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