Skip to content

Telecom R & D

WebRTC , SIP , IMS, VoLTE , SaaS , SBC , REST , Cloud , IOT , media Streams

  • Telecom Info
    • IP Multimedia Subsystem
    • Service Broker
    • Legacy telecom
    • Telecom Architectures
    • Access and Physical Layer
  • SIP
    • Session Initiation Protocol
    • IMS
  • SIP servers
    • Kamailio
    • Opensips
    • freeswitch
    • Oracle SIP server
    • asterisk
  • Media Streams
    • Live Streaming and Broadcasting
    • gstreamer
    • Wowza Media Server
    • Media Processes
  • Kamailio
  • About Me
  • Opensips
  • WebRTC SaaS
  • STUN and TURN
  • WebRTC Media Stack
  • WebRTC security

Tag: webrtc encryption

WebRTC Security Architecture

WebRTC Security Architecture

WebRTC Security Identity Management , Browser Security , Authentication and Media encryption. Browser Threat Model Best practices for the Webrtc comm agents ICE TURN challenges DTLS SRTP

Posted on April 24, 2015March 25, 2025 by altanaiPosted in WebRTC securityTagged browser threat Model, CORS, DTLS, ICE, ICE security, identity management, SCTP, SOP, SRTP, TCB, VOIP security, webrtc encryption. 1 Comment

Categories

  • Access and Physical Layer (8)
  • Auxiliary Technologies for VoIP (8)
    • MOS and Call Quality (1)
    • Natural Language Processing (NLP) (1)
  • Internet of Things (7)
    • Bluetooth Low Energy (1)
    • Raspberry pi (2)
    • RFID (1)
    • Robotics (3)
  • Media Processes (18)
    • codecs (1)
    • gstreamer (1)
    • Live Streaming and Broadcasting (13)
    • Video Analytics (1)
    • Wowza Media Server (2)
  • Protocols (8)
    • XMPP (2)
  • Session Initiation Protocol (SIP) Frameworks (40)
    • JAINSLEE (3)
    • RCS (3)
    • SIP (8)
    • SIP servers (25)
      • asterisk (1)
      • freeswitch (5)
      • Kamailio (11)
      • Opensips (3)
      • Oracle SIP server (1)
    • SIPServlets (1)
  • Signals (2)
  • Telecom Architectures (37)
    • cloud telephony (2)
    • Data Privacy and SIP security in Voice over IP (5)
    • IP Multimedia Subsystem (11)
    • Legacy telecom (3)
    • Service Broker (4)
    • SIP monitoring and Notification (1)
    • Telecom Info (8)
    • VPN (2)
  • Web RealTimeComm. ( WEBRTC) (40)
    • Augmented Reality (2)
    • STUN and TURN (2)
    • TangoFX ( Open Source Conferencing Server) -Archived (3)
    • WebRTC Media Stack (6)
    • WebRTC SaaS (5)
    • WebRTC security (2)
    • WebRTC standards (11)
    • WebRTC usercases and service (8)
    • WHIP and WHEP (1)
https://www.linkedin.com/in/altanai/

WebRTC Integrator’s Guide

Top Posts & Pages

  • Session Initiation Protocol (SIP) Service Creation…
    Session Initiation Protocol (SIP) Service Creation…
  • Hosted IP-PBX and SBC
    Hosted IP-PBX and SBC
  • XMPP Client Server  Setup and Programming
    XMPP Client Server Setup and Programming
  • JANUS as WebRTC SFU
    JANUS as WebRTC SFU
  • 5G and IMS
    5G and IMS
  • IP Multimedia Subsystem (IMS)
    IP Multimedia Subsystem (IMS)
  • Freeswitch PBX system
    Freeswitch PBX system
  • WebRTC Audio/Video Codecs
    WebRTC Audio/Video Codecs
  • SIP ( Session Initiation Protocol )
    SIP ( Session Initiation Protocol )
  • 4G/Long Term Evolution (LTE), VOLTE
    4G/Long Term Evolution (LTE), VOLTE
altanai

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

Verified Services

View Full Profile →

Join 134 other subscribers

Tags

2G 3G 4G Application programming interface Application server Arduino asterisk bea weblogic brekeke Call routing Communications protocol CSP DTLS eclipse FFMpeg freeswitch GSMA H264 HTTP REST ICE IM IMS IN Instant messaging Intelligent Network Intelligent Networks IOT IP address IP Multimedia Subsystem IPSec Jainslee Java JavaScript JSLEE Kamailio kapanga LTE MCU Media server NAT OTT pstn QUIC ramudroid raspberrypi RCS Real-time communication Real-time Transport Protocol RTC RTCP RTMP RTP RTPengine rtsp SBC sdp Security Service-oriented architecture service broker Service harmonization Session Border Controller Session Initiation Protocol Sip sip invite sip server SRTP STUN Telecom Telecom Evolution Telecommunications Telecom Service Provider TFX TURN unified communication VOIP VPN WebRTC Wowza xlite XMPP

Category Cloud

Access and Physical Layer Augmented Reality Auxiliary Technologies for VoIP Data Privacy and SIP security in Voice over IP freeswitch IP Multimedia Subsystem JAINSLEE Kamailio Legacy telecom Live Streaming and Broadcasting Opensips Protocols Raspberry pi RCS Robotics Service Broker Signals SIP SIP servers STUN and TURN TangoFX ( Open Source Conferencing Server) -Archived Telecom Architectures Telecom Info VPN WebRTC Media Stack WebRTC SaaS WebRTC security WebRTC standards WebRTC usercases and service XMPP
September 2026
M T W T F S S
 123456
78910111213
14151617181920
21222324252627
282930  
« Mar    

Recent Comments

L. Garcia's avatarL. Garcia on Performance of WebRTC sites an…
Unknown's avatarAnonymous on NAT traversal using STUN and…
Unknown's avatarAnonymous on VoIP/ OTT / Telecom Solution s…
What is IPTV Player… on IPTV ( Internet Based Televisi…
Unknown's avatarAnonymous on VOIP Call Metric Monitoring an…
Unknown's avatarAnonymous on Proxying Media Streams via Kam…
Unknown's avatarAnonymous on Proxying Media Streams via Kam…
My Tweets
  • LinkedIn
  • Google
  • GitHub
  • YouTube

RSS Feed RSS - Posts

RSS Feed RSS - Comments

RSS Telecom R & D

  • Inside the Telco Backbone: IPSec secures RAN, Core, and Edge in Cloud‑Native Networks
  • OBS Studio + WebRTC: Building and Testing Ultra-Low Latency Streaming
  • IPSec vs MASQUE
  • Digital Rights Management (DRM)
  • Scalable Multicast Media Streaming Protocols
  • Encapsulting Protocols
  • Multihoming protocols and mobility
  • Low Latency Media streaming
  • Fault Tolerance and Error Correction in WebRTC
  • AEC (Echo Cancellation) and AGC (Gain Control) in WebRTC
Website Powered by WordPress.com.
  • Subscribe Subscribed
    Telecom R & D
    Join 51 other subscribers

    Have a WordPress.com account? Log in now.

  • Telecom R & D
    View site in Reader
    Manage subscriptionsSign upLog in
    Report this content
    Collapse this bar
Loading Comments...