Summary:	ASTERISK-25924: chan_pjsip: Polycom SRTP problem
Reporter:	Stacy Vinson (svinson)	Labels:
Date Opened:	2016-04-14 13:14:42	Date Closed:	2017-04-20 00:16:58
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	13.8.0	Frequency of Occurrence	Constant
Related Issues:
Environment:	ubuntu server 14.04 x64 Asterisk 13.8.0 and Asterisk 13.8.1 Polycom VVX 600 US software 5.4.3	Attachments:	( 0) console.txt ( 1) debug_log.txt ( 2) pjsip.conf.txt ( 3) pjsip.conf.txt ( 4) PJSIP-debug.txt
Description:	I'm having a problem getting my Polycom VVX 600 working with SRTP and PJSIP I get a 488 after the RTP/SAVP. The Polycom works fine with SRTP if i use chan_sip, and my AASTRA 57i works fine with PJSIP and SRTP. The Polycom also works fine with PJSIP if i disable SRTP. I also tested a few soft phones with SRTP enabled on PJSIP and did not have any problems. so it looks like it's only a problem with the polycom phones.
Comments:	By: Asterisk Team (asteriskteam) 2016-04-14 13:14:43.397-0500 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: Rusty Newton (rnewton) 2016-04-15 09:54:36.377-0500 Please read through the linked Asterisk Issue Guidelines, remove the excessive debug from the description and attach it to the issue as .txt. Please provide dialplan and SIP channel driver configuration necessary to reproduce the issue - as well as instructional steps for reproduction. By: Stacy Vinson (svinson) 2016-04-15 10:56:25.904-0500 debug and config files By: Rusty Newton (rnewton) 2016-04-25 17:49:08.286-0500 I"m having trouble locating a Polycom VVX 600 for reproduction. Were you able to reproduce with any other polycom phones? By: Rusty Newton (rnewton) 2016-04-25 17:50:26.710-0500 In addition to that question - can you also provide a new debug file following the guide here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information We want to see the VERBOSE, WARNING, NOTICE, DEBUG, ERROR messages integrated with the SIP trace. By: Asterisk Team (asteriskteam) 2016-05-10 12:00:02.051-0500 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: Stacy Vinson (svinson) 2017-04-19 21:08:06.739-0500 still have this issue with Asterisk 14.4.0, i now have a server and Phone setup just for testing. By: Stacy Vinson (svinson) 2017-04-19 21:11:11.555-0500 also i noticed if i  make a call from a softphone "ZOIPER"  to the polycom the call works fine. the issue is only seen when make a call from the polycom. By: Asterisk Team (asteriskteam) 2017-04-19 21:11:11.800-0500 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: Stacy Vinson (svinson) 2017-04-20 00:16:58.610-0500 after updating the polycom vvx 600 to UC Software Version 5.5.1.15937 and setting srtp.require="1" in the polycom config it's working Thanks for your time.