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Istvan

Rating
1484.06 (4,477,279th) 		Reputation
920 (165,874th)
Page: 1 2 3
Title Δ
Is it possible to hide peers' IP from each other when forcing T... 0.00
What might cause this >1000ms lag in webrtc data channel messages? -3.92
Why the value of SIP timer T4 is chosen as 5 seconds? 0.00
WebRTC video recording using RecordRTC showing error in IE and Safari +4.15
"Error: 488: not acceptable here" in officesip (uses ucca... 0.00
In a SIP INVITE Method Getting 200 Response continuosly -0.06
Processing FMTP line above RTP line in a SIP SDP body? 0.00
How to use a JS SIP library using a non-websockets transport? 0.00
How a STUN server behaves when it receives request with same Transa... -3.92
SIPML 5 Client and SipServlets not works Using WSS +0.06
SIP-WebRTC gateway/bridge: Kurento OR openwebrtc OR Intel CS for we... 0.00
Cannot connect with SIP using Asterisk 0.00
Is possible that browser p2p with non-browser(not client-server,not... 0.00
Integrating Asterisk with WebRTC - ground up 0.00
Can I integrate WebRTC-Native Android demo with my kamilio/freeswit... 0.00
How to set FreeSwitch and STUN/TURN when using resiprocate? 0.00
How to enable PRACK(100rel) with Zoiper client 0.00
Choose what apps mute all other sounds on VOIP call in Windows 0.00
What is the difference between Recursive proxy server and Non-Recur... 0.00
webrtc2sip configuration without sipml5 0.00
WebRTC tunelling through 80/443 tcp +4.44
How to remove unnecesary data from SIP Invite in sipML5 0.00
SIP SDP mangling 0.00
Cannot reach SIP server 0.00
Can I use both SIP and P2P for VoIP? 0.00
SIP trunk returns "SIP2/0 513 Message to big". sipjs and... 0.00
VOIP jitter on first call 0.00
what is the session control, network and media information in webrtc 0.00
SIPp - Multiple client SIP Invite to single server - +0.11
SIP over double nat 0.00
Phone call from Java app and receive SDP 0.00
java - initiate VoIP calls through any VoIP Service Provider 0.00
Calculate Mean Opinion Score (MOS) in MySQL +0.18
Transmit Audio/Music Between Phones 0.00
What protocol(s) is used to deliver VOIP to phones 0.00
RaspberryPi - calling your iPhone from your RaspberryPi +4.20
How to get what codec am I using in WebRTC? -3.85
Is there a way to improve audio quality in WebRTC? 0.00
registration process in sip protocol 0.00
Should the android application create SDP by it's own? -4.06
Some seconds delay on starting sending Voip with Android.net.rtp 0.00
Android Digit Grouping 0.00
VOIP client with command line support 0.00
How to handle 4000 sip users and 10000 calls with same ip? -2.01
Call works only one-way , not both-ways -3.95
Android's AEC and Noise Suppressor aren't working on some d... 0.00
Trying to connect iOS app with SIP +0.06
The poor voice quality of my ios voip app 0.00
android start bluetooth sco not routing mic in lollipop 0.00
how to create SIP uri using java.net.URI class? 0.00