As the title of this article suggests I am going to pen my attempts of streaming / broadcasting Live Video WebRTC call to non WebRTC supported browsers and media players such as VLC , ffplay , default video player in Linux etc. Some of the high level archietctures for streaming Webrtc Video to multiple endpoints … Continue reading Streaming / broadcasting Live Video call to non webrtc supported browsers and media players
Author: altanai
Specialized in CPaaS, carrier-grade WebRTC-SIP telecom platforms for Unified communication-collaboration, signaling gateways, SBC, soft turrets, IoT-surveillance and telecom integrations.
Ardent contributor to Open Source software, avid freelancer, innovator and technical writer (telecom.altanai.com ).
Inventor of "RamuDroid" an IOT Road-Cleaning robot
Author of book "WebRTC Integrator's Guide" published by Packt
TFX platform
So I haven't written anything worthy in a while , just published some posts that were lying around in my drafts . Here I write about the main thing . some thing awesome that I was trying to accomplish in the last quarter . << TFX is now live in chrome store , open and … Continue reading TFX platform
Sip server Brekeke
We used Brekeke SIP server to run our SIP applications . Although there are newer versions of Brekeke SIP server out now . More awesome than before , we prefer using the old one for the sake of not messing with legacy SIP applications . The official site for brekeke is - http://www.brekeke.com/sip/ . A general … Continue reading Sip server Brekeke
Steps for building and deploying WebRTC solution
Error in connectivity , errors in console , blank video are the problems that might appear . So well err things begin to get a bit complicated from here . To bypass network firewalls , corporate net policies , UDP blocks and filters we require a TURN server .
WebRTC SDKs Analysis
The fundamental holes in WebRTC specification are still the same with less being done to fulfill them . Ofcourse now there are abundance of interactive WebRTC API each using a new masking method to call the same old WebRTC API function of getusermedia and peer-connection .
Kamailio Architecture, Core and Modules
kamailio has a modular architecture with core components and modules to extend the functionality, this article will be discussing few of the essential modules in Kamailio.
IPTV ( Internet Based Television ) and VOD ( Video on Demand)
IPTVIPTV multicast Mobile TV Video On Demand (VoD)WebRTC for IPTV and VOD We know the power of the Internet protocol suite as it takes on the world of telecom. Already half of the Communication has been transferred from legacy telecom signaling protocols like SS7 to IP based communication ( Skype, Hangouts, WhatsApp, Facebook Messenger, Slack, … Continue reading IPTV ( Internet Based Television ) and VOD ( Video on Demand)
Telephony Solutions with Kamailio
Rich features set suiting to telephony domain that includes IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; Json and XMLRPC control interface, SNMP monitoring. To integrate with a carrier grade telecom network as SBC / gateway / inbound/outbound proxy , it can act as IPv4-IPv6 gateway , UDP/TCP/SCTP/WS translator and even had NAT and anti DOS attack support .
Proxying Media Streams via Kamailio’s RTP Proxy
Kamailio is a SIP server which does not play any role by itself in media transmission path. this behaviour leads to media packets having to attempt to stream peer to peer between caller and callee which in turn many a times causes them to get dropped in absence of NAT management To ensure that media … Continue reading Proxying Media Streams via Kamailio’s RTP Proxy
OTT ( Over the Top ) Communication applications
Market trends are not in favour of Telecom Service /providers with increasing use of OTT ( Over The Top ) applications like WhatsApp, Facebook messenger, Google hangouts, skype, Viber, etc. OTT applications are often blamed to take a stake in voice traffic revenue by using IP calls where the telco could've charged based on its … Continue reading OTT ( Over the Top ) Communication applications
Service Creation Environment (SCE ) for SIP Applications
Develop a SCE ( Service Creation Environment ) to addresses all aspects of lifecycle of a Service, right from creation/development, orchestration, execution/delivery, Assurance and Migration/Upgrade of services.
Call Continuity from Mobile GSM/LTE network to VoIP/WebRTC
Call Continuity from Mobile GSM/LTE network to VoIP/WebRTC
Harmonization of services between generations of telecommunication core layers
Telecommunication service Harmonization Gateways based Harmonization Service Broker based Harmonization Legacy switches vs Softswitches Service Delivery Layer in Legacy vs Harmonized Services A communication system can be made up of many components which are individually undergoing evolution such as access layer generations, and core layer upgrades. Harmonized and uniform open standard-based service delivery platforms over … Continue reading Harmonization of services between generations of telecommunication core layers
WebRTC communication over Web Services
This post is about communication from application to WebRTC using Web Services. For instance showing advertisements on WebRTC interface before p2p streaming or even during. Advertisements could be an overlay or an multiplexed stream. WebRTC + Advertisement Engine HTTP and XML is the basis for Web services. The WebRTC engine, in addition to media stream … Continue reading WebRTC communication over Web Services
OfficeSIP
This post describes the installation , setup and configuration of Office SIP server to provide a registrar to our SIP based WebRTC application .
Telecom R & D 