| Summary: | ASTERISK-28256: Video plays back in slow motion | ||||
| Reporter: | John Bittner (JohnBittner) | Labels: | |||
| Date Opened: | 2019-01-22 21:50:47.000-0600 | Date Closed: | 2019-01-23 05:10:40.000-0600 | ||
| Priority: | Major | Regression? | |||
| Status: | Closed/Complete | Components: | Applications/app_playback Applications/app_voicemail | ||
| Versions: | 13.24.1 16.1.1 | Frequency of Occurrence | Constant | ||
| Related Issues: |
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| Environment: | Centos 6.10 | Attachments: | ( 0) createrecording.pcapng ( 1) msg0000.h264 ( 2) msg0000.txt ( 3) msg0000.wav ( 4) playback.pcapng | ||
| Description: | Have 8845 Cisco video phones setup using chan_sip on asterisk 13 / 16. Video between the phones works perfect, but If one of the phones leaves a message in voicemail, the voicemail system records the video and audio as it should.
On voicemail playback of this voicemail, the video is very slow, almost in slow motion but the audio is fine. Tested the voicemail files using playback and get the same issue. I changed the format to just wav to see if that helped… no change Tested on Asterisk 13 and 16 …. no difference. The phones are set up for ulaw & h264. Also tested with Linphone for Windows. Linphone to 8845 video works perfectly. Linphone to record a video voicemail works. Playback of video voicemail does not work at all on Linphone. I can play back the recorded message from the cisco but again it's slow. | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2019-01-22 21:50:48.689-0600 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]. 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. By: John Bittner (JohnBittner) 2019-01-22 21:51:49.760-0600 Wireshark captures By: John Bittner (JohnBittner) 2019-01-22 21:52:16.645-0600 Voicemail recordings. By: John Bittner (JohnBittner) 2019-01-23 08:50:49.140-0600 While I see you closed this, the other open ticket shows asterisk using PJSIP. I am not, I am using chan_sip. Not sure if this maters much. Also why is the old ticket suspended ? By: Asterisk Team (asteriskteam) 2019-01-23 08:50:49.405-0600 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: Kevin Harwell (kharwell) 2019-01-23 09:12:50.037-0600 [~JohnBittner], for this issue chan_sip vs chan_pjsip shouldn't matter. However if it does there is a link back to this issue to reference the information within. Also the other issue is marked suspended because at some point in the past the issue moved into the suspended state, and for some reason at this time JIRA makes it hard (if possible?) to mark it back as unresolved. | ||||