Summary:	ASTERISK-25065: SRTP failing over time
Reporter:	Sam Ultima (samultima)	Labels:
Date Opened:	2015-05-05 17:38:42	Date Closed:	2015-06-25 12:20:35
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/SRTP
Versions:	13.3.2	Frequency of Occurrence	Occasional
Related Issues:
Environment:	Centos 6.5 (Current version: 6.12.65-27) running 4GB ram & dual CPU's. VOIP PHONES: Polycom IP670	Attachments:	( 0) replication.txt
Description:	From all appearances, SRTP appears to function except over time, calls begin failing. Temp solution is to restart phone and resume calls after registration. Debug shows the following errors: 2015-05-05 15:32:51] WARNING[3055][C-00000006] sdp_srtp.c: Unacceptable a=crypto tag: 13 [2015-05-05 15:32:51] WARNING[3055][C-00000006] chan_sip.c: Rejecting secure audio stream without encryption details: audio 2228 RTP/SAVP 8 0 9 127 SIP/2.0 488 Not acceptable here
Comments:	By: Rusty Newton (rnewton) 2015-05-07 18:40:50.270-0500 Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need: 1. The specific steps or actions you took that caused you to encounter the problem. 2. The behavior you expected and the location of documentation that led you to that expectation. 3. The behavior you actually encountered. To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines [2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Rusty Newton (rnewton) 2015-05-07 18:41:38.295-0500 Please describe step by step how to reproduce the issue and include the above mentioned logs to demonstrate the problems observed. By: Sam Ultima (samultima) 2015-05-07 19:31:02.315-0500 The fastest way we could reproduce problem was to make a conference call (to any number-voicemail in our case) and place it on hold. Once we attempt to retrieve call from hold, there is complete silence/no audio. Further call attempts result in fast busy signal on phone and above error messages. By: Rusty Newton (rnewton) 2015-05-08 15:11:34.107-0500 Hey, I can't reproduce the issue with the information you have here. You'll have to post your configuration (extensions.conf, sip.conf, etc) and demonstrate how to reproduce the issue. The paragraph you have provided is not sufficient. You need to provide step by step instructions and point to logs showing when the issue occurs. Also, as I mentioned previously you'll need to attach logs demonstrating the problem. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information When you have the required information you can press Send Back or Enter Feedback to send the issue back to us. By: Sam Ultima (samultima) 2015-05-08 16:05:00.897-0500 Config & logfiles attached By: Sam Ultima (samultima) 2015-05-08 16:06:08.760-0500 replication instructions attached By: Rusty Newton (rnewton) 2015-05-29 18:44:34.079-0500 reattaching reporter's file "New Text Document.txt" as "replication.txt" By: Rusty Newton (rnewton) 2015-06-01 18:27:08.954-0500 You mentioned that this starts failing over time. When following your replication guide - about how many calls does it take to generally see the failure? Are we talking dozens? Or hundreds? By: Sam Ultima (samultima) 2015-06-01 23:59:59.938-0500 I can usually replicate problem with 3-15 calls. By: Sam Ultima (samultima) 2015-06-16 10:27:01.985-0500 Upgraded asterisk from 6.12.65.27 to 6.12.65-28 & Polycom IP phone from 4.08.1608 to 4.0.9.0509. Have not been able to replicate problem since upgrades. By: Rusty Newton (rnewton) 2015-06-24 18:04:53.564-0500 What do you mean "Upgraded asterisk from 6.12.65.27 to 6.12.65-28" ?  Those Asterisk versions do not exist. By: Sam Ultima (samultima) 2015-06-24 23:53:44.104-0500 This is what appears under admin > system admin > updates: PBX Firmware: 6.12.65-28 PBX Service Pack: 1.0.0.0 By: Rusty Newton (rnewton) 2015-06-25 08:42:52.761-0500 Okay, yeah that is a FreePBX version and not an Asterisk version. FreePBX is a GUI and a distribution that includes Asterisk. Asterisk is the communications engine underneath. Well, since you've upgraded and you can no longer reproduce the issue then I'm going to close this out. However it would be appreciated if you could figure out what Asterisk version you have upgraded from and to so that others will know where this issue exists and where it is possibly fixed. Thanks! By: Sam Ultima (samultima) 2015-06-25 12:14:02.299-0500 13.4.0, however, this may also be a phone issue (note the Polycom IP670 upgrade above) -Sam By: Asterisk Team (asteriskteam) 2015-06-25 12:14:02.726-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Rusty Newton (rnewton) 2015-06-25 12:20:21.692-0500 Thanks!