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sipsorcery

Rating
1481.30 (4,491,634th)
Reputation
23,334 (5,592nd)
Page: 1 2 3 4 ... 11
Title Δ
SIPp: Cancel scenario 487 before 200 0.00
Media Foundation EVR no video displaying +0.03
Validity of SIP ACK response to SIP 200 OK message +0.03
SIP vs RTSP . Which one should be used for video sessions 0.00
Output RTSP stream with ffmpeg 0.00
How to create IMFSample for WindowsMediaFoundation H.264 encoder MFT 0.00
How to tell if a DXVA decoder has fallen back to software decoding 0.00
Get SIP user available status through low-level message 0.00
How to use Media Foundation when application is video source and st... 0.00
IMFSourceReaderCallback::OnReadSample Function 0.00
VoIP-SIP : how to test VoiceMail system with sipp tool +0.06
VoIP Integration in App & Web 0.00
Playback using Raw Data Bytes Frame by Frame 0.00
What will happen in SIP if both parties send INVITE to each other a... +0.03
Record and Playback Video in a byte format 0.00
Media Foundation Frames in Byte Format Run-Time 0.00
Split Contiguous Buffer in RGB Channels (IMFMediaBuffer) 0.00
Play audio from file to speaker with Media Foundation 0.00
Sip call drop after 30seconds 0.00
SIP UAC crash during a call +0.54
VC++ native mutex heap corruption 0.00
The necessity of ACK in INVITE SIP transactions 0.00
Why do we need SIP "100 Trying" response over TCP? -0.46
VoIP - SIP contact filed not completely empty - is RFC compliant? 0.00
SIP: IPSEC vs TLS 0.00
What's the difference between SIP/XMPP for web conferencing and fil... 0.00
Fail over subscribe with SipSorcery 0.00
Does Asterisk 1.8.* support the key exchange RSA and DSS protocols? 0.00
Meaning of "487 Request Terminated" -0.47
Is UPDATE allowed before PRACK is not received from the network to... 0.00
Why is 200 OK retransmission handled by UAS in SIP Protocol? 0.00
Port for sending rtcp receiver reports +0.03
Create SIP address alias using DNS only +0.03
SIP Redirect via Proxy (SIP.js) 0.00
Use Twilio For Sip calls in and out through ONSIP.com 0.00
Encryption/Decryption with SIP TLS 0.00
Jquery Application that is similar to GMail's Web Phone App that is... 0.00
SIP TCP Channel with SIP Sorcery 0.00
Is there an application-agnostic signaling protocol? +0.21
Cisco visual message waiting indicator ( VMWI or MVI ) 0.00
SUBSCRIBE Response using SIP-Sorcery C# 0.00
Asterisk outgoing call fails 0.00
Incoming calls fail Asterisk 0.00
SIP Sever on Amazon EC2 Server 0.00
Router not forwarding packets to other device through STUN mapped p... 0.00
Rails connect to Asterisk and make phone calls +0.10
Sip parser in c# 0.00
Open source solution for SIP based video streaming server? 0.00
calling the PTSN from a browser 0.00
SIP multiple 2** responses 0.00