| Summary: | ASTERISK-28208: chan_pjsip: 183 without SDP followed by 180 does not result in media | ||||
| Reporter: | Michael Maier (micha) | Labels: | pjsip | ||
| Date Opened: | 2018-12-12 14:07:22.000-0600 | Date Closed: | 2018-12-17 06:51:41.000-0600 | ||
| Priority: | Minor | Regression? | No | ||
| Status: | Closed/Complete | Components: | Channels/chan_pjsip | ||
| Versions: | 13.23.1 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Centos 6 | Attachments: | ( 0) missing-ringback ( 1) missing-ringback.pcapng | ||
| Description: | An extension registered at Asterisk 13.23 initiates an external call (pjsip). After the Invite, the
callee (-> ISP) sends a 100 Trying 183 Session Progress (*without* SDP) Asterisk now sends to the extension: 183 Session Progress (*with* SDP) 183 Session Progress (*with* SDP) (really two times) The callee meanwhile sends 180 Ringing (*without* SDP) which is "forwarded" by Asterisk to the extension with 180 Ringing (*with* SDP) The problem: The extension doesn't create a ringback locally, because it most probably expects it to be sent by the callee - but the callee doesn't send anything (not surprising, because there has been no SDP). | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2018-12-12 14:07:24.578-0600 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: Michael Maier (micha) 2018-12-12 14:10:51.148-0600 The pcap trace contains the complete SIP trace as seen by Asterisk: the extension on the left side and the ISP on the right side. By: Richard Mudgett (rmudgett) 2018-12-12 14:19:14.778-0600 We'll need a log showing the dialplan execution for the call. The extra 183 SIP responses could be caused by dialplan. By: Michael Maier (micha) 2018-12-12 14:48:15.117-0600 Well, Joshua already commented on this problem here: http://lists.digium.com/pipermail/asterisk-users/2018-December/293445.html He already seems to know the problem. It's impossible for me to reproduce the problem again today to get the trace you like to see. By: Michael Maier (micha) 2018-12-13 10:52:44.722-0600 Added dialplan trace. By: Joshua C. Colp (jcolp) 2018-12-17 06:43:42.881-0600 The underlying problem is that chan_pjsip does not keep track of whether it has sent a 183 with SDP or not. It should do so, and if told to send a 180 Ringing afterwards it should then produce inband media instead. By: Julien (tigood) 2019-04-09 06:18:26.795-0500 i add, if the first "183 Session Progress (without SDP)" is with or without SDP, Asterisk will send 2 Session Progress... ASTERISK-28185 | ||||