StackRating

An Elo-based rating system for Stack Overflow
Home   |   About   |   Stats and Analysis   |   Get a Badge
Rating Stats for

Istvan

Rating
1484.06 (4,477,279th) 		Reputation
920 (165,874th)
Page: 1 2 3
Title Δ
mediasoup - miss match between PayloadTypes +0.03
webRTC one to one audio call examples 0.00
Why WebRTC chose RTP max packet size to 1200 bytes? 0.00
WebRTC : Can't talk at the same time 0.00
WebSocket handshake error when connection to endpoint 0.00
built SIP client in JAVASCRIPT +4.17
WebRTC in Chrome (JsSIP) and SDP directive "UDP/TLS/RTP/SAVPF&... +0.16
Where should someone start with WebRTC? -3.66
EasyRTC : Noice cancellation not working, can hear my own voice 0.00
How can I set turn (relay) server In Asterisk. (webrtc) 0.00
App not receiving VOIP after x time on iOS 10.3 > +4.01
Why my WebRTC connection doesn't works at some networks? +0.27
WebSocket connection to 'ws://localhost:9090/' failed: Erro... 0.00
Since which java version SHA-256 and SHA256withRSA are supported fo... +0.24
Delphi Firemonkey TWebBrowser use WebRTC in Android, iOS & MacOS 0.00
sending a PCMA file via RTP/RTPC in Java 0.00
Error in websocket secure layer connection in Asterisk 0.00
Asterisk 13.10.0 does not detect user input. DTMF not working 0.00
WebRTC lowest possible atency +1.36
Web application java voice call 0.00
How to add custom video property to WebRTC SDP? -3.78
Is conference call possible with voIP Android +0.19
How to integrate SIP with RTP into android? 0.00
How to detect RTP packets in a PCAP file? +0.24
How many times SDP protocol is passed in a video call 0.00
Is it possible to have Asterisk as the signalling server for WebRTC... -4.08
java.lang.SecurityException: open3: Neither user 10060 nor current... 0.00
Why VoIP RTP has multiple ports instead of single? +0.11
Test WebRTC video call application 0.00
Provisioning Android Device with SIP Account 0.00
Integrating SIP Client (Softphone) as an ActiveX Control on a web-p... 0.00
How to use WebRTC on Android for Real Time Communication? +0.02
Multiple -Re-invites from UA with incremented Cseq Handling -3.73
How to receive audio/video without accepting the autorequest popup... 0.00
Voip application in sleep states 0.00
Cannot establish WebRTC connection (different codecs and payload ty... +0.55
How to make web based call using asterisk rest interface with clien... 0.00
Js SIP code does not work on Web Page 0.00
Group VOIP calling in Android +0.10
Localhost says upgrade required 0.00
Integrating any codec within an android VOIP application 0.00
Re-INVITE during call results with 481 Call Does Not Exist -3.88
"Update Required" show when i on server in webRTC in node... +0.09
How to find out which server has less hops between user or specific... 0.00
WebRTC "ICE Failed" , error 0.00
Web RTC Server is running in localhost only +0.10
Codec used in a webRTC stream +4.14
Failed to load resource: the server responded with a status of 426... -0.04
SIP call video m line rejected with port as 0 0.00
tool to test ICE/STUN/TURN protocol under WebRTC step by step? 0.00