Summary:	ASTERISK-30162: when chan_iax is used to relay calls, no ringing indication is played
Reporter:	Jaco Kroon (jkroon)	Labels:
Date Opened:	2022-07-27 17:07:06	Date Closed:	2023-02-28 07:44:27.000-0600
Priority:	Major	Regression?	Yes
Status:	Closed/Complete	Components:	Channels/chan_iax2
Versions:	18.13.0	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	simplest reproduction: d1 <= SIP => a1 <= IAX/2 => a2 <= SIP => d2. d1 sends INVITE to a1 (to connect with d2). a1 sends back 100 Trying. a1 sends IAX/2 call establishment to a2 a2 acks this. a2 generates INVITE to d2. d2 responds with 100 Trying d2 sends 180 Ringing. a2 sends RINGING indication to a1. a1 drops RINGING indication. This seems to happen because at the time RINGING is sent, no media is established, and as such the frame is dropped due to no compatible media. Issue introduced by 1b62831f2cfe5dcaa519885dd96b645fc05728e7 (as far as I can tell). We do use IAX/2 Jitter Buffer, not sure if this happens without JB. https://gerrit.asterisk.org/c/asterisk/+/18689 was already generated.  This does solve the problem, however, as per feedback, the patch may well be incomplete.
Comments:	By: Asterisk Team (asteriskteam) 2022-07-27 17:07:08.936-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Friendly Automation (friendly-automation) 2023-02-28 07:44:28.451-0600 Change 19919 merged by Friendly Automation: chan_iax2: Fix jitterbuffer regression prior to receiving audio. [https://gerrit.asterisk.org/c/asterisk/+/19919|https://gerrit.asterisk.org/c/asterisk/+/19919] By: Friendly Automation (friendly-automation) 2023-02-28 07:55:18.595-0600 Change 19940 merged by George Joseph: chan_iax2: Fix jitterbuffer regression prior to receiving audio. [https://gerrit.asterisk.org/c/asterisk/+/19940|https://gerrit.asterisk.org/c/asterisk/+/19940] By: Friendly Automation (friendly-automation) 2023-02-28 07:55:46.695-0600 Change 19712 merged by George Joseph: chan_iax2: Fix jitterbuffer regression prior to receiving audio. [https://gerrit.asterisk.org/c/asterisk/+/19712|https://gerrit.asterisk.org/c/asterisk/+/19712]