The use of small cells by mobile operators has been getting a lot of buzz lately. One thing that’s fallen below the radar (no pun intended) is the use of small cells in military systems. In the world of military broadband wireless networks, ambitions have risen in parallel with the ubiquitous deployment of civilian broadband wireless networks. However, the development of advanced military radios has been scaled back in some cases, and manufacturers are under pressure to deliver affordable solutions, quickly and through low-risk programs. To deliver the broadband performance the military needs, at a price governments can afford, some manufacturers have integrated advanced commercial Systems-on-a-Chip (SoCs) designed for civilian small-cell base stations into their military software defined radio (SDR) systems.
Defense OEMs’ pivot towards COTS SDR technologies is understandable – the performance of the latest civilian base station SoCs is spectacular. Not only do these systems implement multiple air interface standards (2G/GSM, 3G/HSPA+, as well as 4G/LTE) and offer very high end-user bitrates (over 10 Mbps), but they also implement advanced adaptive antenna techniques such as 2X2 and 4X4 MIMO. Although small-cell SoCs incorporate many of the features required by tactical radio designers, some SoCs are better suited to military applications than others. In this article, I’ll highlight these differences.
Although most descriptions of WebRTC tout its rich media sharing capabilities, most discussion centers on video and data sharing. As it happens, WebRTC’s audio codec, Opus (at one time called “Harmony”), is among the most remarkable of its features. A free audio codec in development since 2007 by Octasic, Mozilla and the Xiph Foundation in collaboration with Google, Microsoft’s Skype unit and Broadcom, Opus was recently ratified in 2012 by the IETF as RFC 6716. Moreover, after considering ten codecs, the IETF also reached “strong consensus” to adopt Opus as a mandatory-to-implement (MTI) codec for WebRTC.
Update: Since then there has been a push to adopt more legacy codecs to facilitate interworking. See http://tools.ietf.org/html/draft-marjou-rtcweb-audio-codecs-for-interop-00
In any case, to handle the demands of WebRTC, Opus is not your run-of-the-mill codec locked into a specific bitrate and narrowly optimized for speech, high quality music or video. No, Opus can handle everything from low bitrate voice to high bitrate music coding thanks to the fact that it is an amalgam of Skype’s SILK codec that is optimized for low rate voice coding, and Xiph.org’s Constrained Energy Lapped Transform CELT codec, a very low delay, low CPU/memory requirement successor to Vorbis that handles higher bitrates and thus higher quality audio.
When Ericsson officials at the recent AT&T Developer Summit in Las Vegas said that that more innovation is happening in communication today than has happened in the previous 30 years, one of the items they probably had in mind was WebRTC, since their new Web Real Time Communications (WebRTC) development platform enables any device to be reached by an existing mobile number.
Indeed, in the coming weeks you’ll be seeing many WebRTC-related blogs in this space. Why? As Phil Edholm remarked at nojitter.com, “Potentially, WebRTC and HTML5 could enable the same transformation for real-time communications that the original browser did for information.”
Ever since Voice over Internet Protocol (VoIP) was introduced, all calls going to the Public Switched Telephone Network (PSTN) have been plagued with line or electrical echo. I often get asked where these electrical echoes come from and why it is inevitable when using VoIP. We’ve given many seminars on the subject, especially in the Asterisk space where people are suddenly confronted with this problem when building equipment. I’ve put together a basic introduction explaining why echo cancellation is needed when using VoIP over PSTN.
Electrical or line echo is inherent to the PSTN, it has always been present and cannot be avoided because of the nature of the lines connected to homes and offices. These local or subscriber loop circuits use two wires to carry the voice signals while the voice channels use four wires for bi-directional communications beyond the first switch that is the local exchange. The conversion between the four wire and two wire electrical circuits is done by transformer called a telephone hybrid whose goal is to separate the signals’ directions and adapt the impedance of both circuits. Like all analog circuits, the hybrids can’t perfectly match the impedances and that causes part of the signal …
As video becomes one of the most demanding services for network traffic and Internet video becomes an important part of modern life, the quality of experience needs to meet the user’s expectations, regardless of the device or network being used. More and more video is being consumed on smartphones and tablets. The smaller screens allow for lower bit-rates, but the video playback need to start quickly and remain smooth throughout.
Internet Video delivery is challenging because of factors such as high bitrates and sensitivity to delay or packet loss.
In the past, video streaming was typically associated to RTSP, with RTP used for transmission. This protocol uses “VCR-like” commands such as PLAY and PAUSE. In this scheme, the server has to keep track of the client’s state. The server starts playing a stream when giving the PLAY command, and has to maintain the state of each session in order to know what packet to send next. The video stream is based on a single “track”, a file with a fixed encoding profile that cannot change. The quality would quickly suffer when there was a shortage of bandwidth such as congestion leading to packet loss.
Today, HTTP based adaptive …
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Tags2g 3G 4G advanced military radios Apple audio audio codec backhaul basestation broadband bts Cisco civilian codec DSP echo cancellation enterprise G.711 gsm HD video hspa ipad lte military opus codec over the top (OTT) PBX power psychoacoustics rural small cell basestations small cells speech unified communications (UC) video video conferencing video transcoding voip WebRTC wireless