design is based around a Broadcom BCM2835 SoC, which includes an ARM1176JZF-S 700 MHz processor, VideoCore IV GPU, and 512 Megabytes of RAM. The design does not include a built-in hard disk or solid-state drive, instead relying on an SD card for booting and long-term storage. This board is intended to run Linux kernel based operating systems. Generic USB keyboards and mice are compatible with it . It does not come with a real-time clock, so an OS must use a network time server, or ask the user for time information at boot time to get access to time and date info for file time and date stamping. However a real time clock (such as the DS1307) with battery backup can be easily added via the I2C interface.
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
TURN server for WebRTC – RFC5766-TURN , Coturn, Xirsys , Twillio
STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) are protocols that can be used to provide NAT traversal for VoIP and WebRTC. These projects provide a VoIP media traffic NAT traversal server and gateway. TURN Server is a VoIP media traffic NAT traversal server and gateway. This article describes working … Continue reading TURN server for WebRTC – RFC5766-TURN , Coturn, Xirsys , Twillio
NAT traversal using STUN and TURN
STUN and TURN server protocols handle session initiations with handshakes between peers in different network environments . In case of a firewall blocking a STUN peer-to-peer connection, the system fallback to a TURN server which provides the necessary traversing mechanism through the NAT.
continue : Streaming / broadcasting Live Video call to non webrtc supported browsers and media players
This blog is in continuation to the attempts / outcomes and problems in building a WebRTC to RTP media framework that successfully stream / broadcast WebRTC content to non webrtc supported browsers ( safari / IE ) / media players ( VLC )
Streaming / broadcasting Live Video call to non webrtc supported browsers and media players
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
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.
