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Voice/Video Platforms using
MCU/SFU concepts
Agora Platform
• Uses Agora Communications as Service (CaaS)
with their own proprietary audio and video
codec's for P2P and distributed algorithms to
connect calls faster.
• Agora SDK has support for audio, video and chat.
• Device Apps uses Agora own SDK and not werbtc.
• For browser calls, they are using their own wbrtc
for P2P calls and in some scenarios own SDK.
Agora SDK
• Calls are secured with encryption and one
cannot hack the calls using capturing tools
• Agora solution is not Peer to Peer and Server
architecture to ensure voice and video
qualities.
• In some cases, they allow P2P calls and mixed
with Server handling calls
• Platforms supported
– IoS,Android, Windows, MAC OS, Web
Agora SDK measurements
Function Platform Scale
Communication Android, iOS, Windows,
Mac, Web
Audio: 10000 people in
a channel
Video: upto 7 people in
a channel
Live Broadcast Android, iOS, Windows,
Mac
7 video hosts and 10000
audio hosts in a channel
Signalling Android, iOS, Windows,
Mac
Number of Participants:
up to at least 100, 000
people in a channel
Agora SDK Architecture
Agora Cloud
Servers
Agora Cloud Servers
Several data centre
hosting hundreds of
servers
App1
with
Appid1
App2
with
same
Appid1
App3
with
another
Appid2
App4
with
another
Appid
Distributed
and
replicated
database
Agora SDK Architecture
• Several data centre’s in different geo-locations
and hundreds of servers hosting Server
functionality
– Voice, video, chat, group chat and recording
• One channel and multiple users joining audio
and video session
• Selective Forwarding functionality supported
for multiparty conference using unique
application id
Radisys Video conferencing System
• Radisys supports video conferencing for VoLTE, TDM,
VoIP, VoWiFi, and WebRTC endpoints.
• IMS and VoLTE IR.92 and IR.94 specifications
implemented for video conferencing
• IMS MRF product supports video conferencing
• Transcoding done at SBC level
• Radisys has three different products
– Hardware based MPX-1200 supports LTE and voice and
video over LTE
– vMRF deployed in bare metal form and linux distrubutions
– Webconnect – Java based platform
Radisys Voice/Video Conferencing
Janus Webrtc Gateway
• Janus has webrtc gateway similar to our
product supporting below scenarios
– App to App calling media bypassing on server side
– App to PSTN transcoding performed in gateway
– Video room using SFU
Janus Webrtc Gateway
APP1
Janus webrtc
gateway
APP2
PSTN Network
Transacoding
Live streaming
Video rooms
Video room joining
Vide room joining
APP3
App3 calling PSTN
number. Webrtc
gateway does
transcoding
Switch RTC Video conferencing
• Switch RTC (https://switchrtc.com/) built
video conferencing using webrtc SFU
• Video conferencing, live streaming and group
chat functionalities implemented
• Webrtc API’s used for video conferencing
• PSTN calling not supported
SwitchRTC Platform – SFU method
Server is based
on webrtc API’s
APP1
APP2
APP3
APP4
APP5
APP5
• Native Apps and webconnect
Tokbox OpenTok Platform
• OpenTok supports client and Server SDK for voice,
video, char and conferencing services
• Webrtc used for audio and video communication
• Servers configured in cloud platform and unique
session Id shared for video room joining
• Supports IoT using linux based libraries
• Python used to generate session Id’s and client
Id’s.
– Separate SDK API’s used to generate and use client
and session Id’s.
OpenTalk Architecture
Twilio Platform
• Twilio uses webrtc for native apps and browser for 3rd party developers
• Twilio planning to release SFU using webrtc in next release for video
conferencing and video rooms joining (according to their website)
• SFU enables dynamically selecting video tracks and distributing to end
users
• Cloud Platform with multiple servers used for video conferencing
• Features supported
– Video calling, video rooms
– Screen sharing, Video’s sharing, Audio/Video recording
– Peer to Peer media without turn server (DTLS encrypted) when users in same
network. Initially stun/turn and later switched to P2P
– Media through Turn servers when P2P is not possible
– Turn Servers can be configurable in client SDK based on their need
– SFU (upcoming product)
Twilio Cloud Platform
Twilio Cloud
PlatformApp1
with
Appid1
App2
with
same
Appid1
App3
with
another
Appid2
App5
with
another
Appid
App4
with
another
Appid
Inter
communication
TwiLio Server Architecture
APP1
APP2
APP3
APP4
APP5
APP5
Signalling Server – SIP,
websocket, https encrypted
P2P (no media server
involved)
Turn Server (firewalls
traversal)
P2P media
No Server
Media via Turn Server
Signalling always via server
TwiLio Architecture Contd..
Transcoding
• Video Transcoding not supported in current released
• H.265 video codec under implementation
OPUS – G.711 (A/Mu)
SILK – G.711 (A/MU)
Dynamic data
rate
supported in
OPUS codec
VP8,VP9 and H264 codec's
supported
Twilio Audio/Video Conferencing
APP1
APP2
APP3
APP4
APP5
APP5
Audio Mixer
Video SFU
STUN/TURN
P2P media
No Server
Media via Turn Server
Signalling always via server
Proposed Architecture to develop
video conferencing
• App to App scenario
– Both Turn and Direct Peer to Peer solution needs to be implemented
– Call connecting speed needs detailed analysis both on SDP and SIP
signalling level
– Audio and Video is quality is primary goal of this architecture
– Variable bit rate audio codec should be enforced this release
– Video quality and dynamic fps needs to be considered
– Video rooms with dynamically users joining and leaving
• Webrtc Gateway features
– Audio/Video conferencing functionality will be implemented in
webrtc gateway
– Transcoding
• OPUS to PCMA
• Other codec's should be considered
Video Conferencing Server
Architecture
• Need to ensure video codec transparent across devices
• Audio/Video recording and flexible playing recorded clips
• Options for apps to share audio/video clips
• SIP level encryption needed either with TCP/TLS or Websocket to ensure
protection

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Voice video different_platforms_v1

  • 2. Agora Platform • Uses Agora Communications as Service (CaaS) with their own proprietary audio and video codec's for P2P and distributed algorithms to connect calls faster. • Agora SDK has support for audio, video and chat. • Device Apps uses Agora own SDK and not werbtc. • For browser calls, they are using their own wbrtc for P2P calls and in some scenarios own SDK.
  • 3. Agora SDK • Calls are secured with encryption and one cannot hack the calls using capturing tools • Agora solution is not Peer to Peer and Server architecture to ensure voice and video qualities. • In some cases, they allow P2P calls and mixed with Server handling calls • Platforms supported – IoS,Android, Windows, MAC OS, Web
  • 4. Agora SDK measurements Function Platform Scale Communication Android, iOS, Windows, Mac, Web Audio: 10000 people in a channel Video: upto 7 people in a channel Live Broadcast Android, iOS, Windows, Mac 7 video hosts and 10000 audio hosts in a channel Signalling Android, iOS, Windows, Mac Number of Participants: up to at least 100, 000 people in a channel
  • 5. Agora SDK Architecture Agora Cloud Servers Agora Cloud Servers Several data centre hosting hundreds of servers App1 with Appid1 App2 with same Appid1 App3 with another Appid2 App4 with another Appid Distributed and replicated database
  • 6. Agora SDK Architecture • Several data centre’s in different geo-locations and hundreds of servers hosting Server functionality – Voice, video, chat, group chat and recording • One channel and multiple users joining audio and video session • Selective Forwarding functionality supported for multiparty conference using unique application id
  • 7. Radisys Video conferencing System • Radisys supports video conferencing for VoLTE, TDM, VoIP, VoWiFi, and WebRTC endpoints. • IMS and VoLTE IR.92 and IR.94 specifications implemented for video conferencing • IMS MRF product supports video conferencing • Transcoding done at SBC level • Radisys has three different products – Hardware based MPX-1200 supports LTE and voice and video over LTE – vMRF deployed in bare metal form and linux distrubutions – Webconnect – Java based platform
  • 9. Janus Webrtc Gateway • Janus has webrtc gateway similar to our product supporting below scenarios – App to App calling media bypassing on server side – App to PSTN transcoding performed in gateway – Video room using SFU
  • 10. Janus Webrtc Gateway APP1 Janus webrtc gateway APP2 PSTN Network Transacoding Live streaming Video rooms Video room joining Vide room joining APP3 App3 calling PSTN number. Webrtc gateway does transcoding
  • 11. Switch RTC Video conferencing • Switch RTC (https://switchrtc.com/) built video conferencing using webrtc SFU • Video conferencing, live streaming and group chat functionalities implemented • Webrtc API’s used for video conferencing • PSTN calling not supported
  • 12. SwitchRTC Platform – SFU method Server is based on webrtc API’s APP1 APP2 APP3 APP4 APP5 APP5 • Native Apps and webconnect
  • 13. Tokbox OpenTok Platform • OpenTok supports client and Server SDK for voice, video, char and conferencing services • Webrtc used for audio and video communication • Servers configured in cloud platform and unique session Id shared for video room joining • Supports IoT using linux based libraries • Python used to generate session Id’s and client Id’s. – Separate SDK API’s used to generate and use client and session Id’s.
  • 15. Twilio Platform • Twilio uses webrtc for native apps and browser for 3rd party developers • Twilio planning to release SFU using webrtc in next release for video conferencing and video rooms joining (according to their website) • SFU enables dynamically selecting video tracks and distributing to end users • Cloud Platform with multiple servers used for video conferencing • Features supported – Video calling, video rooms – Screen sharing, Video’s sharing, Audio/Video recording – Peer to Peer media without turn server (DTLS encrypted) when users in same network. Initially stun/turn and later switched to P2P – Media through Turn servers when P2P is not possible – Turn Servers can be configurable in client SDK based on their need – SFU (upcoming product)
  • 16. Twilio Cloud Platform Twilio Cloud PlatformApp1 with Appid1 App2 with same Appid1 App3 with another Appid2 App5 with another Appid App4 with another Appid Inter communication
  • 17. TwiLio Server Architecture APP1 APP2 APP3 APP4 APP5 APP5 Signalling Server – SIP, websocket, https encrypted P2P (no media server involved) Turn Server (firewalls traversal) P2P media No Server Media via Turn Server Signalling always via server
  • 18. TwiLio Architecture Contd.. Transcoding • Video Transcoding not supported in current released • H.265 video codec under implementation OPUS – G.711 (A/Mu) SILK – G.711 (A/MU) Dynamic data rate supported in OPUS codec VP8,VP9 and H264 codec's supported
  • 19. Twilio Audio/Video Conferencing APP1 APP2 APP3 APP4 APP5 APP5 Audio Mixer Video SFU STUN/TURN P2P media No Server Media via Turn Server Signalling always via server
  • 20. Proposed Architecture to develop video conferencing • App to App scenario – Both Turn and Direct Peer to Peer solution needs to be implemented – Call connecting speed needs detailed analysis both on SDP and SIP signalling level – Audio and Video is quality is primary goal of this architecture – Variable bit rate audio codec should be enforced this release – Video quality and dynamic fps needs to be considered – Video rooms with dynamically users joining and leaving • Webrtc Gateway features – Audio/Video conferencing functionality will be implemented in webrtc gateway – Transcoding • OPUS to PCMA • Other codec's should be considered
  • 21. Video Conferencing Server Architecture • Need to ensure video codec transparent across devices • Audio/Video recording and flexible playing recorded clips • Options for apps to share audio/video clips • SIP level encryption needed either with TCP/TLS or Websocket to ensure protection