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SRTP – Secure Real-time Transport Protocol | Adaptive …
https://www.adaptivedigital.com/secure-rtp/
SRTP: Secure Real-time Transport Protocol. SRTP is a security profile for RTP that adds confidentiality, message authentication, and replay protection to that protocol. SRTP is ideal for protecting Voice over IP traffic because it has no effect on voice quality and payload overhead is only minimally affected.
FreePBX 12 Update issue - FreePBX - FreePBX Community Forums
https://community.freepbx.org/t/freepbx-12-update-issue/27482
[2015-02-23 09:04:27] WARNING[4310][C-00000e1b] chan_sip.c: We are requesting SRTP for audio, but they responded without it! We never have been able to get Jitsi to work with encryption but it has worked without encryption enabled up to this past week. Consequently encryption is turned off on the device entry of this extension and Jitsi has ...
Make Asterisk use RTP even when SRTP is requested ...
https://community.asterisk.org/t/make-asterisk-use-rtp-even-when-srtp-is-requested/46155
Does anyone know how to make 11.9 revert to RTP even if asked to use SRTP?? Here is the message I get: WARNING[1155][C-0000001e]: chan_sip.c:10543 process_sdp: We are requesting SRTP for audio, but they responded without it! Many Thanks!!!-John
user_srtp - Snom Service Hub
https://service.snom.com/display/wiki/user_srtp
DESCRIPTION. Your phone supports RTP encryption via SRTP. If you want to encrypt your outgoing audio (RTP) stream, this option must be “on”. Both parties have to enable the RTP Encryption option to establish an SRTP call. RTP encryption has nothing to do with SSL/ TLS. The keys are sent in the SDP part of SIP messages.
One way audio after hold when using SRTP - Poly …
https://community.polycom.com/t5/VoIP-SIP-Phones/One-way-audio-after-hold-when-using-SRTP/td-p/97479
The other side of the call always hears the caller though, and we also hear the both parties in the recorded call. To replicate it: - make a call where Phone is using SRTP. - during the duration of the call the SRTP sequence number exceeds maximum value of 0xffff (65535) and starts again from 0. -after this, the polycom phone goes on hold then ...
Need help undertending SRTP log - Asterisk Support ...
https://community.asterisk.org/t/need-help-undertending-srtp-log/48792
You are using SIP. When a SIP client tries to establish a session, it sends data using session description protocol. That is what is used to request SRTP. In order to have SRTP, that SDP must specify an encryption key to be used for the media. en.wikipedia.org/wiki/Session_De … n_Protocol
What does this mean? - General Help - FreePBX Community Forums
https://community.freepbx.org/t/what-does-this-mean/17436
[2013-04-19 12:04:23] WARNING[1644][C-00000000]: chan_sip.c:10452 process_sdp: We are requesting SRTP for audio, but they responded without it! Is this a configuration issue with the SOMA-320? What specifically is wrong with its set-up or what configuration change in FreePBX will allow it to work?
Securing Internet Telephony Media with SRTP and SDP
https://tools.cisco.com/security/center/resources/securing_voip.html
This technical white paper is a practical guide for applying SRTP to voice, fax, and other IP telephony media. It is intended for engineers and gives an overview of IP telephony security and technical fundamentals of SRTP. Network administrators can use this paper to learn how to configure SRTP and SDP security services for various purposes.
How to decrypt SRTP from WebRTC audio call? - VoIP Forum ...
https://community.spiceworks.com/topic/2265775-how-to-decrypt-srtp-from-webrtc-audio-call
From the other Forums, some says - by being a passive observer - it is possible to decrypt the SRTP if we have the server RSA key whereas some says it may not possible to decrypt DTLS-SRTP. I am stuck without any luck and badly in need of finding a solution to decrypt the SRTP. I have couple questions here, any inputs would be highly appreciated.
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