Summary:	ASTERISK-27674: chan_sip: RTP framing issues on outgoing calls
Reporter:	Jean Aunis - Prescom (PrescomJA)	Labels:
Date Opened:	2018-02-14 03:02:23.000-0600	Date Closed:	2018-03-08 14:58:41.000-0600
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Channels/chan_sip/General
Versions:	14.7.4 15.2.1	Frequency of Occurrence	Constant
Related Issues:
Environment:		Attachments:	( 0) full_case_1.txt ( 1) full_case_2.txt
Description:	When placing outgoing calls with chan_sip, RTP packetization time is not honoured, whether it is received from the remote endpoint in an SDP "ptime" attribute, or it is forced by configuration in the "allow" line. *Case 1 : ptime received from the remote endpoint* Given I have configured a peer with "autoframing=yes" When the peer replies to an INVITE with an SDP containing "a=ptime:40" Then RTP is still sent with a packetization time of 20 ms *Case 2 : packetization configured in sip.conf* Given I have configured a peer with "allow=alaw:40" When the peer replies to an INVITE without specifying "ptime" Then RTP is still sent with a packetization time of 20 ms Please note that the outgoing INVITE contains the proper "ptime" attribute in the SDP, the only problem is the RTP packetization.
Comments:	By: Asterisk Team (asteriskteam) 2018-02-14 03:02:24.218-0600 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: Asterisk Team (asteriskteam) 2018-02-14 03:02:25.564-0600 The module you are reporting the issue against is no longer supported as a core module but your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available. Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1] If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way. [1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2]: http://www.asterisk.org/community/discuss [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties By: Jean Aunis - Prescom (PrescomJA) 2018-02-14 03:04:02.401-0600 Adding log full for both cases. By: Friendly Automation (friendly-automation) 2018-03-08 14:58:43.676-0600 Change 8208 merged by Joshua Colp: chan_sip: Fix improper RTP framing on outgoing calls [https://gerrit.asterisk.org/8208|https://gerrit.asterisk.org/8208] By: Friendly Automation (friendly-automation) 2018-03-08 15:55:07.798-0600 Change 8453 merged by Jenkins2: chan_sip: Fix improper RTP framing on outgoing calls [https://gerrit.asterisk.org/8453|https://gerrit.asterisk.org/8453] By: Friendly Automation (friendly-automation) 2018-03-08 17:14:19.352-0600 Change 8461 merged by Jenkins2: chan_sip: Fix improper RTP framing on outgoing calls [https://gerrit.asterisk.org/8461|https://gerrit.asterisk.org/8461]