|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|
|Versions:||SVN 220.127.116.11 10.12.2||Frequency of|
|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.
* 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.
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
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.