kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2026-05-04 20:33:53 +00:00
Code Issues Packages Projects Releases Wiki Activity

rtp: Enable srtp replay protection

Browse Source 
Add option "srtpreplayprotection" rtp.conf to enable srtp
replay protection.

ASTERISK-29260
Reported by: Alexander Traud

Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458
This commit is contained in:
Alexander Traud
2021-01-26 10:09:53 -07:00
committed by George Joseph
parent 7d15655f9d
commit 389b8b0774
5 changed files with 43 additions and 4 deletions
Download Patch File Download Diff File Expand all files Collapse all files

12
configs/samples/rtp.conf.sample
View File

@@ -45,6 +45,18 @@ rtpend=20000
; connected. This option is set to 4 by default.
; probation=8
;
; Enable sRTP replay protection. Buggy SIP user agents (UAs) reset the
; sequence number (RTP-SEQ) on a re-INVITE, for example, with Session Timers
; or on Call Hold/Resume, but keep the synchronization source (RTP-SSRC). If
; the new RTP-SEQ is higher than the previous one, the call continues if the
; roll-over counter (sRTP-ROC) is zero (the call lasted less than 22 minutes).
; In all other cases, the call faces one-way audio or even no audio at all.
; "replay check failed (index too old)" gets printed continuously. This is a
; software bug. You have to report this to the creator of that UA. Until it is
; fixed, you could disable sRTP replay protection (see RFC 3711 section 3.3.2).
; This option is enabled by default.
; srtpreplayprotection=yes
;
; Whether to enable or disable ICE support. This option is enabled by default.
; icesupport=false
;