Dissecting a VoIP application

Dissecting a VoIP Application on Prezi

Comments

  1. Reliable VOIPSeptember 29, 2021 at 1:12 AM

    Great article by the great author, it is very massive and informative but still preaches the way to sounds like that it has some beautiful thoughts described so I really appreciate this article. Best small business phone systems canada service provider.

    ReplyDelete
  2. The VOIP Guru, Inc.May 3, 2022 at 3:30 AM

    I liked your work and the way in which you have shared this blog here about Voip Phone System For Small Business It is a beneficial and helpful blog for us. Thanks for sharing a blog like this.

    ReplyDelete
  3. NancyNovember 14, 2022 at 2:08 AM

    You are now prepared to place and answer calls. Your address book, which contains the names and phone numbers of people you frequently call, is stored on a SIM card. Non-steering IoT SIM Card

    ReplyDelete
  4. NancyNovember 16, 2022 at 7:20 AM

    When you provide your visitors with useful information, search engines will also appreciate it. because the useful material on your site might boost its popularity. SEO consultant philadelphia

    ReplyDelete

Post a Comment

Popular posts from this blog

Bandwidth Estimation in WebRTC (and the new Sender Side BWE)

Image
Bandwidth estimation is probably the most critical component in the video engine of WebRTC. The bandwidth estimation (BWE) module is responsible for deciding how much video* traffic you can send without congesting the network to prevent degradation of the video quality. In the past bandwidth estimation algorithms used to be very rudimentary and based mostly on packet loss. Basically we used to start increasing slowly the video bitrate until we detected packets being lost.  To detect the packet loss you use the standard RTCP feedback mechanisms where the receiver side reports packet loss periodically using RTCP Receiver Report (RR) messages. Modern bandwidth estimation algorithms are more advanced and try to detect congestion before it is bad enough to make routers discard packets. These algorithms predicts congestion analyzing the delay between packets. The idea is that when you start having some congestion, the buffers in the routers will start filling and the delay will be
Read more

Improving Real Time Communications with Machine Learning

Image
When we talk about the applications of Artificial Intelligence / Machine Learning (AI/ML) for Real Time Communications (RTC) we can group them in two different planes: Service Level : There are many features that can be added to a videoconference service, for example identification of the participants, augmented reality, emotion detection, speech transcription or audio translation .  These features are usually based on image and speech recognition and language processing. Infrastructure Level : There are many ways to apply ML that do not provide new features but improve the quality and/or reliability of the audio/video transmission. Service level applications are fun, but they are more for Product Managers and I like technology more, so in the next sections I will try to describe possible applications of AI/ML for Real Time Communications at Infrastructure Level organizing those ideas in five different categories. Optimizing video quality Some of the ML algorithms used fo
Read more

Controlling bandwidth usage in WebRTC (and how googSuspendBelowMinBitrate works)

Image
There are cases when we would like to limit the maximum bitrate being transmitted by WebRTC to avoid wasting resources in the user endpoints or save money reducing the bandwidth usage in our servers.   This is because the maximum bitrate by default in Chrome is around 2Mbps and for many use cases a much lower bitrate provides still pretty good quality.   BTW using a lower bitrate can also help with stability of quality in multiparty scenarios by reducing the amount of competition among different streams. There is no simple API to configure the maximum bitrate in WebRTC (although there is one in ORTC) but there are 3 ways to do this by mangling the SDP. 1. U se the standard  b=AS:BITRATE (Chrome) or b=TIAS:BITRATE  (Firefox) attributes in the SDP for the audio or video channel[1] 2. Use codec specific attributes (this work at least for opus audio codec with maxaveragebitrate property) [2] 3. Use the proprietary x-max-bitrate attribute in the video channel of the SDP answer.  For e
Read more