| Summary: | ASTERISK-25451: Broken video - erased rtp marker bit | ||
| Reporter: | Stefan Engström (StefanEng86) | Labels: | |
| Date Opened: | 2015-10-07 08:50:31 | Date Closed: | 2015-10-07 12:29:45 |
| Priority: | Major | Regression? | |
| Status: | Closed/Complete | Components: | Resources/res_rtp_asterisk |
| Versions: | 13.5.0 13.6.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | For this particular issue, an attempted resolution https://gerrit.asterisk.org/#/c/1388 was created before this jira issue.
When making a call between two sip peers that have video, and when using intermediate local channels in the call's chain of channel connections (possibly even without), the RTP marker bit on the outgoing RTP video packets is always set to 0, which causes issues in the receiving sip client's video decoding process. This issue probably affects many more versions of asterisk. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2015-10-07 08:50:33.695-0500 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. 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: Richard Mudgett (rmudgett) 2015-10-07 12:29:45.493-0500 Patch merged into master and v13 branches. | ||