Summary: | ASTERISK-29171: codec list in INVITE is reversed when using remote | ||||
Reporter: | Mark Petersen (roadkill) | Labels: | |||
Date Opened: | 2020-11-20 05:54:41.000-0600 | Date Closed: | |||
Priority: | Minor | Regression? | Yes | ||
Status: | Open/New | Components: | Resources/res_pjsip_sdp_rtp Resources/res_pjsip_session | ||
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 ( 1) consol.log ( 2) extensions.conf ( 3) pjsip_wizard.conf ( 4) pjsip.conf | ||
Description: | when using the incoming_call_offer_pref=remote the order of INVITEs what is send out the the order is reversed, as it do not match the one in the config
INVITE from A with G722,PCMA,PCMU INVITE to B the order is reversed PCMU,PCMA,G722 (order is reversed!) 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 05:54:42.902-0600 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 06:11:26.732-0600 What is the specific scenario for this? What is the Asterisk console log and dialplan for it? By: Mark Petersen (roadkill) 2020-11-20 06:36:38.819-0600 I have attached copy of the extensions.conf pjsip_wizard.conf pjsip.conf but it is basically just a SBC sitting on our public interface and sending traffic between our provider and internal servers so there is one context that dial the interna server By: George Joseph (gjoseph) 2020-11-20 09:29:37.802-0600 I'm working on finishing Advanced Coded Negotiation now. By: Mark Petersen (roadkill) 2021-04-29 06:13:40.005-0500 if you need any help testing let me know |