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Rating Stats for

Rajesh

Rating
1494.40 (4,294,583rd) 		Reputation
515 (266,833rd)
Page: 1
Title Δ
change video resolution in sip mid call 0.00
How to connect with SIP server for VoIP call in iOS? +0.03
How to dump sip traces according to ip in separate files? 0.00
what is the user of ice-options in ICE Protocol? -0.35
Which ICE candidate am I using and why? -0.12
How to modify answer SDP for record audio only with Kurento? -3.89
Asterisk sends SIP bye packet to wrong destination 0.00
expire=0 in multiple contact REGISTER + SIP 0.00
P-CSCF is adding an extra paramater ( i=1) in the via header field... 0.00
SIP Trunk messaging 0.00
SIP protocol / call waiting -0.03
Does SIP RFC allows multiple endpoints registered to one account? +4.01
receiving service from SIP without using a server +0.06
How it's work Kamailio transformations +0.04
VoIP Setup with Softphones and Twilio backend 0.00
Convert voip audio to text for debugging -4.03
Backend for WoIP on WebRTC, asterisk? -4.04
Securing RTP packets without encrypting each packets 0.00
Mobicents SIP error response handling - what's the proper way t... -0.06
MySQL installation on Kamailio 0.00
freeswitch and sip.js how to configure websocket 0.00
What is the difference between Twilio Elastic SIP Trunking, and Twi... 0.00
what sip middleware can I use to modify sip headers 0.00
How to modify the handling of a SIP message in Freeswitch? +4.02
How to connect VOIP to PSTN? -3.96
How to understand that UDP Packet is carrying SIP message -3.19
What is the minimal SDP answer to receive WebRTC Audio and Video? +0.00
SIP UAC crash during a call -2.20
wireshark capture sip traffic and save an XML file with a specific... 0.00
Making calls via internet in android -0.09
How is SIP scaled for high load? -0.43
Online Multimedia Streaming and Two Way communication / Conference... 0.00
Why do we need SIP "100 Trying" response over TCP? +3.70
How to decode SIP packets? -4.42
Android VOIP Encryption 0.00
Locate answer SDP packet in Wireshark 0.00
Presence Server working details 0.00
Intercept SIP outgoing packets and alter them enroute 0.00
Asterisk 12.6 // TURN // WebRTC // No audio +3.98
What is the use of from-tag in SIP request? +1.96
SIP codec negotiation -0.11
Linux SIP Client just to get incoming number 0.00
Freeswitch and webRTC: media rejected with 488 +4.00
Freeswitch and webRTC: media rejected with 488 -4.00
Asterisk call dropped after 32 seconds -0.14
How to figure out when SIP call is started -0.12
Meaning of "487 Request Terminated" +3.76
Diameter protocol. What's the expected behaviour if several CERs ar... 0.00