[asterisk-bugs] [Asterisk 0017855]: Asterisk with SRTP Record and Playback Issue
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Sep 1 13:10:31 CDT 2010
The following issue has been CLOSED
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https://issues.asterisk.org/view.php?id=17855
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Reported By: hemanshurpatel
Assigned To:
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Project: Asterisk
Issue ID: 17855
Category: Resources/res_srtp
Reproducibility: always
Severity: block
Priority: normal
Status: closed
Target Version: 1.8.0
Asterisk Version: Older Addons - please test a newer version
JIRA: SWP-2050
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: open
Fixed in Version:
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Date Submitted: 2010-08-13 01:25 CDT
Last Modified: 2010-09-01 13:10 CDT
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Summary: Asterisk with SRTP Record and Playback Issue
Description:
I am using asterisk successfully with SRTP for Calling.
I havent tested my installation for record and playback since i added SRTP
support to it.
i have configured extention 999 for record task:
exten => 999,1,Set(_SIPSRTP_CRYPTO=enable)
exten => 999,n,Set(_SIPSRTP=1)
exten => 999,n,Answer
exten => 999,n,Record(hemu.ulaw)
exten => 999,n,Hangup
When i call 999, application receives INVITE packet with crypto tag in
SDP. I am using Grandstream GVX3410 phone which supports SRTP.Apllication
process the message, Process_crypto function call and call is established.
I mean asterisk starts recording, but while recording in rtp_recvfrom
function, rtp->srtp == NULL and so it doesnt decrypt the packets. Stores
packet as it is.
and same happens while playback, and so packet which asterisk sends while
playback are encyrpted with tags of recording call, so its just blank
screen and Noise on speakers.
What extra happens in normal call is that asterisk also processes 200 OK
with crypto header and so it calls process_crypto function two times while
in record and playback function it calls those function one time only.
Anyone has got same problem?
regards
Hemanshu Patel
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(0126530) twilson (administrator) - 2010-09-01 13:10
https://issues.asterisk.org/view.php?id=17855#c126530
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Please try using the 1.8 branch or trunk and then apply the patch attached
to the srtp.diff on issue 17563 (also posted to reviewboard at
https://reviewboard.asterisk.org/r/878/). The combination of newer code and
that patch should fix the issue that you are experiencing.
I'm going to go ahead and close this as a duplicate of 17563. We can
discuss there as it is kind of a merge of a couple of SRTP issues that
should be adressed now.
Issue History
Date Modified Username Field Change
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2010-09-01 13:10 twilson Note Added: 0126530
2010-09-01 13:10 twilson Status feedback => closed
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