Summary:	ASTERISK-26223: chan_sip: Changes to Encryption option not accepted on a reload of chan_sip
Reporter:	Sebastian Gutierrez (sum)	Labels:
Date Opened:	2016-07-20 11:30:23	Date Closed:	2016-10-28 15:10:14
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:	13.9.1	Frequency of Occurrence
Related Issues:	is related to ASTERISK-26313 chan_sip : Asterisk restart seems to be required for changing encryption option
Environment:		Attachments:	( 0) full.log ( 1) sip1.conf ( 2) sip2.conf
Description:	changing on the file and reloading sip Encryption from *yes* to *no* is not taking the change although "sip show channels" shows Encryption=no but this log is shown and the call is not possible, restarting asterisk make it work. chan_sip.c:10715 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio Added from a comment: Below, the steps to reproduce : * I have a working peer, configured as in sip1.conf * I delete the parameters related to WebRTC as in sip2.conf * I do a SIP reload * When I try to make a call I get the following message in the asterisk CLI : {code}WARNING[5107][C-00000004] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio{code} Attached is the asterisk full log with the trace of the failed call with the SIP debug. callid asterisk : C-00000004 CallID SIP : f51acc7c-7893-4f17-bde9-36865e50cf9f After a restart of asterisk, line working is properly.
Comments:	By: Asterisk Team (asteriskteam) 2016-07-20 11:30:24.699-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2016-07-21 06:24:34.065-0500 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: Asterisk Team (asteriskteam) 2016-08-04 12:00:01.737-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines By: benasse (benasse) 2016-08-23 05:04:16.750-0500 Below, the steps to reproduce : * I have a working peer, configured as in sip1.conf * I delete the parameters related to WebRTC as in sip2.conf * I do a SIP reload * When I try to make a call I get the following message in the asterisk CLI : {code}WARNING[5107][C-00000004] chan_sip.c: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio{code} Attached is the asterisk full log with the trace of the failed call with the SIP debug. callid asterisk : C-00000004 CallID SIP : f51acc7c-7893-4f17-bde9-36865e50cf9f After a restart of asterisk, line working is properly. By: Sebastian Gutierrez (sum) 2016-10-27 09:53:38.871-0500 ping? this issue should not be closed. By: Asterisk Team (asteriskteam) 2016-10-27 09:53:39.032-0500 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) 2016-10-28 08:53:46.762-0500 This seems to be a duplicate of ASTERISK-26313. I'm not sure we need both of these open.. Please help me understand if there is a difference between ASTERISK-26223 and ASTERISK-26313. By: Sebastian Gutierrez (sum) 2016-10-28 09:39:17.957-0500 yes its the same as far as I can see. By: benasse (benasse) 2016-10-28 12:53:31.074-0500 It the same, I opened another issue because I could not reopen this one. By: Rusty Newton (rnewton) 2016-10-28 15:10:14.744-0500 Thanks. I'm going to close this out so we can continue tracking the issue in ASTERISK-26313