Author Archives: Dimitri Osler

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About Dimitri Osler

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RTP, RTCP and Jitter Buffer

jitter bufferIn 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

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Media transfer in Unified Communications – SDP protocol

SDP-protocolMedia 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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Introducing BOSH and WebSocket Transport protocols

BOSH and WebSocket protocolsThis 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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SIP and XMPP standards in Unified Communications

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.

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Web, Hybrid or Native Mobile Apps

Web, Hybrid or Native Mobile apps – which one to choose? In the previous blog article we discussed web-based communication solutions and their advantages over native ones. This time we will discover why it is more preferable to use native or hybrid applications for mobile devices.

Mobile devices like smartphones and tablets have two important limitations:

  • CPU and battery usage
  • Technology limitations

Because these two factors must be taken into account, collaboration apps for mobile phones must still be native or hybrid apps. Native means using Objective-C and Swift (with iOS devices) and Java (with Android devices). Hybrid means that a mix of native code and Webviews are included in the app.

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