Summary:	ASTERISK-27892: No audio with IPV6 enabled ENDPOINT
Reporter:	Purchasing Agent (hawkeye)	Labels:	pjsip
Date Opened:	2018-06-01 15:22:46	Date Closed:	2018-06-01 17:06:18
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	13.19.1	Frequency of Occurrence	Constant
Related Issues:
Environment:	FreePBX 14 / Asterisk 13.19.1	Attachments:	( 0) pjsip-configs.zip
Description:	When making or getting calls where my endpoint is using IPv6 there is no audio. Since my extension records all calls, I played back the recording and there is most definetly audio. The problem is audio is not being sent to the IPv6 enabled endpoint. When endpoint uses IPv6, rtp is incorrectly binding to ipv4 port instead of ipv6 port on PBX. For debugging, added rtp_ipv6=yes to one extension in /etc/asterisk/pjsip.endpoints.conf and restarted asterisk the endpoint now shows: rtp_ipv6                           : true Logging of a call from the router in front of PBX shows: 15:39:57 firewall,info FROMPHONE forward: in:br-management out:ether2, src-mac 00:18:74:27:82:40, proto UDP, [2607:fea8:20df:fc31: 20b:82ff:fe95:223a]:5012->[2607:f938:1001::b154:4edb]:16376, len 190 ran netstat -tupan while making a call and it shows the port 16376 is udp4 and not udp6 udp        0      0 xxx.193.49.61:16376     0.0.0.0:*
Comments:	By: Asterisk Team (asteriskteam) 2018-06-01 15:22:47.351-0500 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: Richard Mudgett (rmudgett) 2018-06-01 15:35:12.240-0500 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: Purchasing Agent (hawkeye) 2018-06-01 16:11:55.454-0500 Richard, what more info do you need?? By: Purchasing Agent (hawkeye) 2018-06-01 16:13:38.349-0500 There is no errors generated in cli> or log files. We've done a lot of digging to show rtp is using ipv4 when connected to ipv6 endpoint. Make a call, no audio. Not sure what else you want to see,. By: Joshua C. Colp (jcolp) 2018-06-01 16:21:35.096-0500 That isn't enough information. Packet capture, actual full configuration, a description of the scenario, information about the network. Without those there's nothing we can do. By: Purchasing Agent (hawkeye) 2018-06-01 17:03:37.285-0500 PJSIP config files from FreePBX Server with the audio issues By: Joshua C. Colp (jcolp) 2018-06-01 17:06:18.572-0500 This is not a bug. This is configuration. Each endpoint has the following set: {noformat} media_address=104.193.49.61 bind_rtp_to_media_address=yes {noformat} So RTP is being bound to the given IPv4 address. Removing these would cause it to bind to both IPv4 and IPv6 instead on all addresses. By: Richard Mudgett (rmudgett) 2018-06-01 17:20:31.437-0500 Reattached config files without the auth sections. By: Purchasing Agent (hawkeye) 2018-06-02 23:37:32.687-0500 Rmugett, you are correct. We have figured out how to get freepbx to not include the media_address line for the endpoints. Thank you for your trouble shooting. By: Asterisk Team (asteriskteam) 2018-06-02 23:37:33.106-0500 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: Purchasing Agent (hawkeye) 2018-06-02 23:38:39.850-0500 Rmugett, you are correct. We have figured out how to get freepbx to not include the media_address line for the endpoints.