| Summary: | ASTERISK-29172: codec list in 200/183 is not using the common denominator | ||||
| Reporter: | Mark Petersen (roadkill) | Labels: | |||
| Date Opened: | 2020-11-20 05:59:25.000-0600 | Date Closed: | 2020-11-20 09:31:18.000-0600 | ||
| Priority: | Minor | Regression? | Yes | ||
| Status: | Closed/Complete | Components: | pjproject/pjsip | ||
| Versions: | 18.1.0 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | CentOS Linux release 8.2.2004 (Core) Linux 4.18.0-193.19.1.el8_2.x86_64 x86_64 | Attachments: | ( 0) codec.pcap | ||
| Description: | when using the outgoing_call_offer_pref=remote the list in the 200/183 is not filtered to only contain what is the common denominator
INVITE from A with G722,PCMA,PCMU INVITE to B the order is reversed PCMU,PCMA,G722 183 from B PCMA,PCMU (no G722) 183 to A PCMU,PCMA,G722 (G722 is not removed!) endpoint/allow=!all,alaw,g722,ulaw endpoint/incoming_call_offer_pref=remote endpoint/outgoing_call_offer_pref=remote ;remote - Include all codecs in the remote list that are also in the local list preserving the remote order. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2020-11-20 05:59:26.162-0600 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. 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: Joshua C. Colp (jcolp) 2020-11-20 06:09:00.282-0600 This is not currently supported. The functionality to allow the information to traverse back to the calling side has not yet been implemented in PJSIP and is being worked on. By: Asterisk Team (asteriskteam) 2020-11-20 06:24:44.535-0600 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: Mark Petersen (roadkill) 2020-11-20 07:09:10.432-0600 hmm I thought that it only was the codec_prefs_XXXX that was not implemented could we then add a note in the config that it is not implemented or remove it as "codec_prefs_XXXX" has been in (ASTERISK-29109) https://gerrit.asterisk.org/c/asterisk/+/15040/4/configs/samples/pjsip.conf.sample By: Asterisk Team (asteriskteam) 2020-11-20 07:09:10.694-0600 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: George Joseph (gjoseph) 2020-11-20 09:31:18.934-0600 I'm currently working on finishing up Advanced Codec Negotiation so I'm gonna close this as a duplicate but leave it related to ASTERISK-29171. | ||||