| Summary: | ASTERISK-19995: Recording calls is strongly degraded when using RTP packetization of 60 ms (g729: 60) | ||
| Reporter: | Artem Kalatsey (aakalacey) | Labels: | |
| Date Opened: | 2012-06-14 01:26:14 | Date Closed: | 2012-07-23 19:06:41 |
| Priority: | Major | Regression? | Yes |
| Status: | Closed/Complete | Components: | |
| Versions: | 1.8.13.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | CentOS 5.6 or CentOS 5.8 on different hardware | Attachments: | ( 0) test_for_forum.7z |
| Description: | When configuring RTP packetization, setting the timer to 60 ms (allow=g729: 60 in sip.conf). Heavily degraded call recording of a remote client, although the voice does not undergo any changes. When decreasing the timer recording quality become better, with a 20 ms difference in the voice channel is disappear. All works on real hardware, as the writer use mixmonitor. The problem is relevant for both version 1.6 and for 1.8 Looked at different forums - any ideas to remedy is not found. | ||
| Comments: | By: Matt Jordan (mjordan) 2012-06-22 08:41:39.696-0500 We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information By: Rusty Newton (rnewton) 2012-07-23 19:06:30.545-0500 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||