Summary: | ASTERISK-21901: speex16 call to app_record with wav format results in a playable, but horrible sounding audio file | ||
Reporter: | Peter Katzmann (pk16208) | Labels: | |
Date Opened: | 2013-06-12 02:55:28 | Date Closed: | 2017-12-19 09:15:56.000-0600 |
Priority: | Major | Regression? | |
Status: | Closed/Complete | Components: | Applications/app_record Codecs/codec_speex |
Versions: | SVN 1.8.20.2 10.12.2 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | ubuntu linux | Attachments: | ( 0) 73.wav ( 1) 76.wav ( 2) record-failled.log ( 3) record-ok.log ( 4) speex16err.pcap ( 5) speextoRecord.txt ( 6) speextovoicemail.txt |
Description: | [Edit by RNewton: speex16 to app_record results in a bad sounding audio recording. speex works fine. See details in comments and attachments.]
A call is forwarded from server b to server a fro voicemail recording codec is speex16 recording sound like a drunken robot asterisk cli is very unresponsive/slow during recording when if force speex or ulaw/alaw is codec recording is good. | ||
Comments: | By: Peter Katzmann (pk16208) 2013-06-12 02:57:21.807-0500 Debug with working recording By: Rusty Newton (rnewton) 2013-06-25 17:44:26.393-0500 Let's clarify what you are saying so I can know where to look for the issue. Do these two points sum up the issue? 1. SIP <-(negotiated speex16)-> Asterisk (recording) : results in bad recording, plus CLI unresponsiveness during recording. 2. SIP <-(forced speex16)-> Asterisk (recording) : results in a good recording If that is the case.. that would be very odd. Also: * please provide a PCAP (SIP/RTP) that matches your Asterisk DEBUG logs (VERBOSE level 5, DEBUG level 5). By: Peter Katzmann (pk16208) 2013-06-26 01:18:16.161-0500 Your point 2 of the sum up is wrong. Right one: 2. SIP <(forced [speex|ulaw|alaw])> Asterisk (recording) : results in a good recording and responsive cli By: Peter Katzmann (pk16208) 2013-06-26 03:06:22.732-0500 Requested pcap, call to *96 is the one with the voicemail recording By: Rusty Newton (rnewton) 2013-06-26 10:07:50.328-0500 bq. 2. SIP <(forced [speex|ulaw|alaw])> Asterisk (recording) : results in a good recording and responsive cli Did you mean "speex" or "speex16" there? By: Peter Katzmann (pk16208) 2013-06-26 10:09:55.887-0500 Yes i always stating speex for normal speex and speex16 for wb speex By: Rusty Newton (rnewton) 2013-07-01 15:39:53.519-0500 Reproduced in SVN-branch-1.8-r391778 (from this afternoon) Attaching two files from reproduction. Files include CLI"core show channel" output and shell "# file <filename>" output for two scenarios for interesting comparison: * call from jitsi (speex16) to Asterisk(sip user set to speex16 only) and app_record (with options ".wav,,,k") * call from jitsi (speex16) to Asterisk(sip user set to speex16 only) and app_voicemail (voicemail.conf configured for wav49|gsm|wav) Both result in recordings. The recording resulting from app_record is playable but sounds horrible and is mostly unintelligible. The recording resulting from app_voicemail is playable and sounds fine. See respective files for details speextoRecord.txt speextovoicemail.txt Same test with app_record using speex (not speex16) results in a fine sounding audio file. By: Joshua C. Colp (jcolp) 2017-12-18 11:58:34.132-0600 Have you experienced this in Asterisk 13 as well? By: Peter Katzmann (pk16208) 2017-12-19 09:09:26.184-0600 I canno't reproduce it between two asterisk 13 servers By: Joshua C. Colp (jcolp) 2017-12-19 09:15:56.282-0600 I shall mark this as suspended then. If you encounter it again you can comment and it will automatically reopen. |