DRM empowers content owners and distributors to successfully manage ad sales rights, produce accurate inventory forecasts, optimize for maximum revenue, deliver yield-optimized ads, and analyze video business performance. Why DRM Still Matters Piracy tooling now targets low-latency and browser-delivered streams, not just VOD assets. Modern stacks blend real-time WebRTC, near-real-time LL-HLS/DASH on QUIC, and traditional … Continue reading Digital Rights Management (DRM)
Tag: WebRTC
Low Latency Media streaming
Low latency is imperative for use cases that require mission-critical communication such as the emergency call for first responders, interactive collaboration and communication services, real-time remote object detection etc. Other use cases where low latency is essential are banking communication, financial trading communication, VR gaming etc. When low latency streaming is combined with high definition (HD) quality, the complication grows tenfold. This article discusses RTMP, RTSP, LL HLS, MPEG-DASH and, WebRTC, SRT as technologies to provide low-latency streaming. It also discusses the TradeOff of Latency vs. Quality and congestion control to avoid packet loss which is detrimental to low latency.
Fault Tolerance and Error Correction in WebRTC
Fluctuating NetworksDynamic Bandwidth estimation JitterBuffer SDP renegotiationDemand for High Quality Video Tradeoff between Latency vs Quality Layering for adaptive streamingBetter compression algorithms vs CPU computeFull INTRA-frame Request (FIR)Picture Loss Indication (PLI)Redundant Encoding (RED) in Media Packets CongestionFeedback Loop Overcome congestion with lower bitrate Reduce frame quality and resolution Congestion control Algorithms : Google Congestion Control … Continue reading Fault Tolerance and Error Correction in WebRTC
AEC (Echo Cancellation) and AGC (Gain Control) in WebRTC
Acoustic EchoHybrid / Electronic Echo in PSTN phonesNoise Suppression in WebRTCEcho CancellationWebRTC Echo CancellationAutomatic Gain Control (AGC) Echo is the sound of your own voice reverberating. If the amplitude of such a sound is high and intervals exceed 25 ms, it becomes disruptive to the conversation. Its types can be acoustic or hybrid. Echo cancellers … Continue reading AEC (Echo Cancellation) and AGC (Gain Control) in WebRTC
SCTP (Stream Control Transmission Protocol)
SCTP is a reliable, message-oriented transport layer protocol. It was initially developed for telephony signaling (specifically for SS7 (Signaling System No. 7)) over IP, which requires robust and reliable messaging, but is widely used in WebRTC for data channels today as a standard. Being suitable for high availability , fault tolerant networks it is a … Continue reading SCTP (Stream Control Transmission Protocol)
TeleMedicine and WebRTC
The solution enables doctors / nurses / medical practitioners and patients to do
- High definition Audio/video calls
- End to end encrypted p2p chats
- Integration with HMS ( hospital management system ) to fetch history of the patients
- Screens sharing to show reports without transferring them as files
- Include more concerned people of doctors using Mesh based peer to peer conferencing feature.
WebRTC Audio/Video Codecs
WebRTC Video CodecsVP8VP9H264/AVC constrainedAV1 (AOMedia Video 1)Stats for video based media stream trackNon WebRTC supported Video codecs H.263H.265 / HEVCWebRTC Audio CodecsOpusAACG.711 (PCMA and PCMU)G.722iLBCiSACSpeexAMR-WB DTMF and 'audio/telephone-event' media typeStats for Audio Media trackDataChannelStats for Datachannel Codecs signifies the media stream's compession and decompression. For peers to have suceesfull excchange of media, they need a common … Continue reading WebRTC Audio/Video Codecs
WebRTC APIs
WebRTC (Web Real-Time Communication) provides a set of JSEP APIs that enable real-time peer-to-peer (P2P) communication in web browsers. These handle media capture, encoding, network traversal (ICE), and secure data transmission. This article describes those in detail and also highlights usecases around varying states of these APIs. MediaDevices getUserMedia()getDisplayMedia()enumerateDevices()getSupportedConstraints()MediaStream and MediaStreamTrackPeer-to-peer connectionsRTCPeerConnection InterfaceCreateOffer() CreateAnswer()Offer/Answer Options - VoiceActivityDetectionRTCSessionDescriptionProfile … Continue reading WebRTC APIs
JavaScript Session Establishment Protocol (JSEP) in WebRTC handshake
SEP is used during signalling via w3c's recommended RTCPeerConnectionAPI interface to set up a multimedia session. The multimedia session description specifies the critical components of setting up a session between local and remote such as transport ports, protocol, profiles. It also handles the interaction with the ICE state machine.
WebRTC CPaaS ( Communication Platform as a Service )
CPasS ( communication platform as a service ) is cloud based communication platform that provides real time communication capabilities. This should be easily integrable with any given external environment or application of the customer, without him worrying about building backend infrastructure or interfaces .
RTCP Reports and QoE metric calculation
RTCP works alongside RTP to monitor and control media streams with QoS feedback, synchronization and session management . This writeup describes the key format and functions of this protocol RTCP (Real-Time Transport Control Protocol ) RTCP Control and ManagementGathers statistics on media connectionSR: Sender Report RTCP PacketRR: Receiver Report RTCP PacketSDES: Source Description RTCP PacketBYE: … Continue reading RTCP Reports and QoE metric calculation
Kamailio WebRTC SIP Server
Why Kamailio?Kamailio SIP Signaling ProxySystem ArchitectureTLS module Configuration GuideStep 1: Create Certificate Directory StructureStep 2: Generate Certificate Authority (CA)Step 3: Generate Server CertificateStep 4: Install CertificatesStep 5: Create CA Certificate ListStep 6: Configure TLS in kamailio.cfgTesting TLS ConfigurationWebsocket moduleConfigurationNAT Detection and HandlingRTPengineInstallationRunning RTP EngineKamailio RTP Engine IntegrationTesting RTP EngineJSSIPJSSIP WebRTC client for KamailioSIP over WEBSOCKET … Continue reading Kamailio WebRTC SIP Server
Session Border Controller (SBC) for WebRTC
SBC became important part of comm systems developed over SIP and MGCP. SBC offer B2BUA ( Back to Back user agent) behavior to control both signalling and media traffic.
Setting up ubuntu ec2 t2 micro for webrtc and socketio
Setting up a ec2 instance on AWS for web real time communication platform over nodejs and socket.io using WebRTC . Primarily a Web Call , Chat and conference platform uses WebRTC for the media stream and socketio for the signalling . Additionally used technologies are nosql for session information storage , REST Apis for getting sessions details to third parties.
RamuDroid
Bot to clean roads and outdoors for a better and cleaner India. It lifts up small objects like plastic cups,wrappers,leaves etc. The droid also provides real-time camera stream and detects obstruction to re-route itself. It can communicates over GSM ,wifi and BLE . It can also be remote navigated via browsers or android. Working : … Continue reading RamuDroid
Telecom R & D 