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Rating Stats for

Sasi Varunan

Rating
1493.81 (4,316,831st) 		Reputation
1,460 (111,724th)
Page: 1 2
Title Δ
How to check whether if I'm running the code on a GCP / AWS ins... 0.00
KaiOS Spatial navigation +0.03
WebRTC in Chrome (JsSIP) and SDP directive "UDP/TLS/RTP/SAVPF&... -0.16
record audio from browser and visualize in real time 0.00
Plivo getting UUID of call recording +0.10
WebRTC: Can I create datachannels with the same label? 0.00
What's missing in Answer SDP (From web browser to android device) +4.15
WebRTC. The video does not show in <video> tag 0.00
Websocket open error, websocket register error 0.00
webrtc remote video on firefox not work +0.16
Recording audio and streaming to clients from nodejs 0.00
Change video input of a video capture card on a web application 0.00
Remote video is black screen or blank in WebRTC 0.00
How to install WebRTC in fusionpbx 0.00
I need to know the STUN server hardware requirements for 10 agents... 0.00
Quickblox Webrtc Audio Calls - Can't hear sound 0.00
Is it possible to use webRTC cross-browser? -0.02
MediaTrack detect highest level +0.54
Using jsSIP in A Project 0.00
I implemented Android SIP sample to audio and video calling but get... +0.08
JsSIP: User is logged out when browser refreshed 0.00
PLIVO SMS Sender Id Issue +0.16
Connecting a Lua client to a Socket.io NodeJS server? +4.20
Is it possible to access a live audio stream in the browser? +0.23
Catch audio stream in freeswitch 0.00
Html5 player customization 0.00
webRTC in node.js -0.00
Analyze sound stream without duplicating it 0.00
Regarding hardware requirements for freeswitch 0.00
How to execute shell script using php? +0.11
google login php with specific domain 0.00
socket.io is not working with static file routing node.js 0.00
webrtc per to per video chat but only need one side to send video t... 0.00
Asterisk call recording not showing any logs and file 0.00
I am getting 404 with socket.io How to resolve this +2.14
One to Many Microphone Streaming Implementation -3.79
FreeSwitch outgoing call. Answering machine(fax) detection 0.00
Initiating call and receiving call in web browser using freeswitch 0.00
Why do I get 'WebSocket opening handshake was canceled' try... 0.00
Freeswitch command to add sip users -0.05
Freeswitch: play ringing tone to A-leg, while we play some message... +4.20
Can i make voice call from android clent to sip js web app 0.00
how change G722 to PCMA codec? 0.00
Is it possible to have Asterisk as the signalling server for WebRTC... +4.08
Freeswitch detect Fax programmatically 0.00
How to put the session.id of caller SIP.JS (wss-binding) to CDR log... +0.26
SipJS and Freeswtch : Not able to receive phone call 0.00
FreeSWITCH filter registrations -3.89
FreeSWITCH originate_timeout timer 0.00
Freeswitch handle pstn calls for switched off numbers +4.27