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
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 (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:
Every day we use the Internet countless times; we:
- buy and sell products;
- book tickets/hotels;
- book appointments for any type of services;
- seek for jobs;
- read movie/book reviews;
- ask for references/ recommendations when want to go on vacation, etc.
The Web serves as a so-called “Door Opener” as it has a great promotional power, opening up new possibilities to Business Owners, on their way to meeting potential customers and their needs.
But how can you make your business “visible” among your competitors on the local level?
How can you make it recognizable among local customers?
The answer is literally on the surface- “Go to the Web,” use Online Directories that allow you to list your business for free and be searched by the customers on an equal basis with the other companies from your area. Continue reading
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.
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.
In this blog article we will discuss the basic standards used for real-time communications — SIP and XMPP — what is the difference, how each of them works, and, which one to choose.
SIP and XMPP
The IETF has two documented standards for real-time communications that are widely implemented: SIP and XMPP.
These standards transport text information and rely on other standards for the actual media transmission.
As both support real-time communications, many question which solution is most suited to their needs.
Let’s briefly explore the history and purpose of both.