Summary:	ASTERISK-12022: Weird noise with Speex codec
Reporter:	Diego Viola (diegoviola)	Labels:
Date Opened:	2008-05-14 21:22:05	Date Closed:	2008-11-21 09:40:44.000-0600
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Codecs/codec_speex
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) 1210821917-SIP-diego.viola-158a4340-out.ogg
Description:	Hello everyone- I just tried the speex codec and every time I make a call I hear too much noise, my libogg is 1.1.3, speex is 1.2beta3, asterisk is 1.4.19.2, compiled with gcc 4.1.2, on Linux 2.6.18, CentOS release 5. I will post some audio files soon... Thanks, Diego
Comments:	By: Diego Viola (diegoviola) 2008-05-14 21:23:50 I forgot to mention, I tried this with softphones, X-Lite and Zoiper, using SIP. Both running on Linux. By: Diego Viola (diegoviola) 2008-05-14 21:59:16 One more note, I tried with other codecs (gsm, ulaw/alaw, etc) and I don't hear that weird noise with this codecs, just with speex. By: Diego Viola (diegoviola) 2008-05-14 22:17:32 Just added the sound file, I recorded it with Monitor() on the server, the server has good (~ 4mbps) internet connection. By: Jason Parker (jparker) 2008-05-19 15:49:13 Does changing the version of libspeex change anything? What is the source codec (from the softphone to Asterisk)? How is this file getting from speex to ogg format? By: Diego Viola (diegoviola) 2008-05-19 15:58:18 >> Does changing the version of libspeex change anything? I tried with speex 1.0.x before, and same thing... that's why I tried speex 1.2beta3 later, which gave me the same results. >> What is the source codec (from the softphone to Asterisk)? Speex. >> How is this file getting from speex to ogg format? I recorded it on the server with Monitor() which saved it to a wav file, then I re-encoded that with oggenc. I played the wav file and had those noises too, I'm 100% sure of this, couldn't post the wav file because it exceeded the file size limit. By: Jason Parker (jparker) 2008-05-19 16:15:14 It might be useful to catch the RTP traffic on the wire with wireshark, and have it save the recording for playback. I'm not too well versed on wireshark, but I know it supports that - could you try doing that? By: Tilghman Lesher (tilghman) 2008-06-03 11:45:51 If you set DONT_OPTIMIZE in 'make menuselect' and recompile, does the noise go away? By: Diego Viola (diegoviola) 2008-06-03 13:58:39 Nope, same thing... I just did that but the noise is still there. I will post my RTP traffic soon... By: Diego Viola (diegoviola) 2008-06-03 14:04:45 I also noticed that when I speak with other people using the Speex codec in my endpoint, I don't get that noise at all. The only times I get that noise is on music-on-hold. I also already updated asterisk to 1.4.20.1 but the problem is still there, this version feels more stable though. By: Tilghman Lesher (tilghman) 2008-06-03 14:08:37 What format is your MOH presently using?  If you change your MOH to uncompressed wav (8000Hz, single channel) format, does the noise go away? By: Diego Viola (diegoviola) 2008-06-03 15:11:38 This is what I have in my menuselect, so I guess my MOH is WAV by default. [*] 1.  MOH-FREEPLAY-WAV [ ] 2.  MOH-FREEPLAY-ULAW [ ] 3.  MOH-FREEPLAY-ALAW [ ] 4.  MOH-FREEPLAY-GSM [ ] 5.  MOH-FREEPLAY-G729 [ ] 6.  MOH-FREEPLAY-G722 By: Tilghman Lesher (tilghman) 2008-06-08 18:10:03 FreeSwitch is irrelevant, as it is a completely different codebase and is not related to Asterisk at all.  You might as well have said that NetMeeting doesn't exhibit this problem. By: Diego Viola (diegoviola) 2008-06-12 14:05:34 Upgraded to 1.4.21 and the problem is still there. By: Tilghman Lesher (tilghman) 2008-06-17 16:08:48 Someone would need to figure out what the actual problem is, first. By: gavin (gavin) 2008-06-22 05:37:44 At first this sounds diffrent than my issue, but after the music on hold starts, it sounds similar.  http://bugs.digium.com/view.php?id=12443 By: Leif Madsen (lmadsen) 2008-11-21 09:40:43.000-0600 I'm going to say this is most likely related to the issue re: GSM and gcc versions. If someone can provide the appropriate information to move this issue along, please feel free to find a bug marshall in #asterisk-bugs on irc.freenode.net to have this issue reopened. Thanks!