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From: daniel h. <dan...@to...> - 2004-11-08 17:36:35
|
Michael Van Donselaar a =E9crit : >>-----Original Message----- >>From: iax...@li...=20 >>[mailto:iax...@li...] On=20 >>Behalf Of daniel huhardeaux >>Sent: Saturday, November 06, 2004 11:41 PM >>To: iaxclient-list >>Subject: Re: [Iaxclient-devel] Next to Audio Caller problem >> >>Michael Van Donselaar a =E9crit : >> >> =20 >> >>>>As sad at the begining of my mail: is working. To summarize, taken=20 >>>>only audio problem in account: >>>> >>>>Diax097a, iaxcomm 02/2004, IaxPhone 0.1.0 and 0.2.0 (all windows),=20 >>>>tkphone (linux) are working. >>>> =20 >>>> >>>> =20 >>>> >>>There are only two changes to the iaxcomm code between Feb=20 >>> =20 >>> >>04 and Oct 04: >> =20 >> >>>1) Feb code had a null pointer function reference when right=20 >>> =20 >>> >>clicking=20 >> =20 >> >>>on the tray icon. This could not affect linux port, as it is in a=20 >>>block that is ifdefed out >>> >>>2) Clicking Dial now clears the extension combobox. >>> >>>Any other changes in behavior you are seeing are not iaxcomm=20 >>> =20 >>> >>specific. >> =20 >> >>>=20 >>> >>> =20 >>> >>Two questions: >> >>1. why iaxcomm windows from october segfault? >> =20 >> > >I strongly suspect that it has something to do with the codec negotiatio= n >code. Other than the two changes noted above, there have been no change= s to >iaxcomm, only (many!) changes to iaxclient. > >I see in another message from Dan, the author of DIAX, that he is gettin= g >complaints about ulaw audio quality. I also noticed that ulaw was menti= oned >in on of the error messages you got. > >Since you are compiling yourself already, and I think I recall you prefe= r >iLBC, try this: Add=20 > >iaxc_set_formats(IAXC_FORMAT_ILBC, IAXC_FORMAT_ILBC); > >right after=20 > >iaxc_start_processing_thread(); > >in main.cc (approx line 206). This will force an iLBC connection. Sinc= e >you have hardware for both platforms, I'd be interested to see how this >works for you. > =20 > Ok, I got it: problem is echo cancelation. If it's activated I have my=20 audio problems. I remain my test schema (echo test from FWD): iaxComm - asterisk - internet - FWD Iax Concerning codecs, FWD allow *only* Ulaw. My * box accept in this order=20 (ilbc,gsm,ulaw,alaw). In iaxComm: iLBC: ok but quality is not so good (choppy) GSM: perfect but I only hear called party, my recorded voice is never=20 coming back (or sended?) ULAW: perfect but ... echo and over 6 sec delay. If on my * box I authorize g729, no audio at all. Waiting your comments if any. > =20 > >>2. I test again with iaxcomm linux from februar: I have audio=20 >>with distortion >> =20 >> > >But did I understand correctly that you got good results with >iaxcomm-win-20040228.zip? > =20 > Yep > =20 > >>What I also notice, is that from time to time iaxcomm is not=20 >>exiting properly. Closing the main window has no action.=20 >>Going to terminal from where iaxcomm was launcheg and ctrl/c.=20 >>Nextime iaxcomm is started, the lock file is deleted. >> >>I suggested some time ago to call someone from you so you=20 >>will be able to hear what happends. Still actual ;-) >> =20 >> > >And as I responded, I do not have any hardware to make this practical. = I >have only one linux box with a sound card, and it is in the basement wit= hout >any speaker or microphone. > =20 > Sorry, didn't saw your message :-[ I could also call an windows=20 client. Anyway, now we got it :-) --=20 Daniel |
From: Dan <dt...@fx...> - 2004-11-08 06:00:59
|
Hi Steven, >----- Original Message ----- >From: "Steven Sokol" <ss...@so...> To: "Iaxclient-Devel" <iax...@li...>; "Asterisk User List" <ast...@li...> Sent: Monday, November 08, 2004 5:59 AM Subject: [Iaxclient-devel] Bad audio on IAX Phone <--> Cisco 7960 Call? >I seem to have hit a problem with calls between users on IAX Phone and > the popular (and expensive) Cisco 7960. The audio quality is TERRIBLE. > Broken up, ragged sounding. It makes no difference which codecs I use > on either end (this is an experimental version of IAX Phone with > multiple codecs). The same problem seems to occur with iaxComm and > Firefly as well. Strange, becaus I use this configuration for more than a year with very good results. The sound is perfect, using GSM on the iaxclient and ulaw on the 7960. The same for ATA186. > > Can anybody else out there with a Cisco 7960 set up and test this? > Here's my current configuration: > > Cisco 7960 w/ SIP Image 7.1.00 > Asterisk Stable-1.0.1 > IAX Phone (any version) or iaxComm soft phone I have version 5 of the SIP iamge and no need to upgrade it... > > Please let me know if you've seen this before or if you can reproduce it. > Unfortunately (or not?..:-) ), I cannot reproduce this. Maybe it is a problem of the firmware version. Best regards, Dan |
From: Dan <dt...@fx...> - 2004-11-08 05:56:55
|
Hi Steve, >----- Original Message ----- >From: "Steve Kann" <st...@st...> > Personally, I prefer speex, but I suspect that I'd lose in a vote. For me , the best result when using DIAX from a notebook with a wireless connection (802.11b), at a marginal level, GSM provides still the best sound. iLBC is alittle bit choppy. Best regards, Dan |
From: Dan <dt...@fx...> - 2004-11-08 05:51:56
|
Hi Steve, >----- Original Message ----- >From: "Steve Kann" <st...@st...> > > On Nov 7, 2004, at 2:49 PM, Dan wrote: > >> Hi, >> >> Try to set the environment variable (even from inside your app, like >> in DIAX): >> PA_MIN_LATENCY_MSEC >> A value of 60 seems to be the best for me (I think that the default is >> 400). >> >> Steve, what about including this in the iaxclient library, as default? > > I could do that; I guess I haven't yet been convinced that this isn't > going to cause problems when machines are heavily loaded. Then a separate function to set this woukd be great. For me the result is spectacular. The delay is now just as small as between two ATAs. > > >> The same for the current used codec. > > I'm not sure what you mean here. I won't make iLBC the default, > because it is not free. Not this. I mean to include a function who will return the codec used in the current call, as it is not allways the default one. This can be done in an existing function too, as an optional parameter. > > Both of these things could be made options configurable from the > Makefile, though. Is ok. Best regards, Dan |
From: Dan <dt...@fx...> - 2004-11-08 05:47:07
|
Hi Steve, >----- Original Message ----- >From: "Steve Kann" <st...@st...> > There should be four different audio situations, but you're only > reporting the result in two cases: > > Call from iaxclient to ATA, with iLBC on the iaxclient side, and > asterisk translating: > Case 1: Audio from iaxclient to ATA. > Case 2: Audio from ATA to iaxclient. The sound is good in bith situations. > > Call from ATA to iaxcleint, uLaw all the way > Case 3: Audio from iaxclient to ATA The sound is perfect. > Case 4: Audio from ATA to iaxclient. The sound is choppy with high latency (seconds) > > Which audio cases result in problems? > Just the decode part of the ulaw codec. > In those cases, you should take a look at the network traffic via > ethereal, and see if the times and timestamps being sent into iaxclient > make sense. For uLaw, you should get frames about every 20ms, and the > timestamps should be separated by 20ms. For iLBC, it should be 30ms > with the way asterisk works. I'll take a look at this today. Best regards, Dan > > |
From: Adam H. <ad...@te...> - 2004-11-08 05:04:39
|
Strange that * would create the correct packets. ethereal the SIP side of a pure G.711 call (IAX g.711 also), are the RTP frames around 20ms apart and timestamps incrementing in 320 lots (or is it 160) Good luck :) -Adam Steven Sokol wrote: > Actually the IAX Phone end sounds fine. Only the Cisco side sounds > bad. No jitter buffer enabled. > > Adam Hart wrote: > >> Both ends? any differences in ethereal? jitterbuffer off? >> >> Steven Sokol wrote: >> >>> I seem to have hit a problem with calls between users on IAX Phone >>> and the popular (and expensive) Cisco 7960. The audio quality is >>> TERRIBLE. Broken up, ragged sounding. It makes no difference which >>> codecs I use on either end (this is an experimental version of IAX >>> Phone with multiple codecs). The same problem seems to occur with >>> iaxComm and Firefly as well. >>> >>> It ONLY appears to happen with the Cisco. My other SIP phones (a >>> Grandstream BT101, a Uniden, an ACT, a Sipura and an ariaVoice) all >>> work just fine with the incoming audio stream from an IAX Client device. >>> >>> It also appears to be a problem with iaxClient calls, as IAX2 calls >>> Asterisk-to-Asterisk sound fine. >>> >>> Can anybody else out there with a Cisco 7960 set up and test this? >>> Here's my current configuration: >>> >>> Cisco 7960 w/ SIP Image 7.1.00 >>> Asterisk Stable-1.0.1 >>> IAX Phone (any version) or iaxComm soft phone >>> >>> Please let me know if you've seen this before or if you can reproduce >>> it. >>> >>> Thanks, >>> >>> Steve >>> >>> P.S. - A multi-codec version of IAX Phone will be out soon. >>> >> >> >> >> ------------------------------------------------------- >> This SF.Net email is sponsored by: >> Sybase ASE Linux Express Edition - download now for FREE >> LinuxWorld Reader's Choice Award Winner for best database on Linux. >> http://ads.osdn.com/?ad_id=5588&alloc_id=12065&op=click >> _______________________________________________ >> Iaxclient-devel mailing list >> Iax...@li... >> https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > |
From: Steven S. <ss...@so...> - 2004-11-08 04:57:46
|
Actually the IAX Phone end sounds fine. Only the Cisco side sounds bad. No jitter buffer enabled. Adam Hart wrote: > Both ends? any differences in ethereal? jitterbuffer off? > > Steven Sokol wrote: > >> I seem to have hit a problem with calls between users on IAX Phone >> and the popular (and expensive) Cisco 7960. The audio quality is >> TERRIBLE. Broken up, ragged sounding. It makes no difference which >> codecs I use on either end (this is an experimental version of IAX >> Phone with multiple codecs). The same problem seems to occur with >> iaxComm and Firefly as well. >> >> It ONLY appears to happen with the Cisco. My other SIP phones (a >> Grandstream BT101, a Uniden, an ACT, a Sipura and an ariaVoice) all >> work just fine with the incoming audio stream from an IAX Client device. >> >> It also appears to be a problem with iaxClient calls, as IAX2 calls >> Asterisk-to-Asterisk sound fine. >> >> Can anybody else out there with a Cisco 7960 set up and test this? >> Here's my current configuration: >> >> Cisco 7960 w/ SIP Image 7.1.00 >> Asterisk Stable-1.0.1 >> IAX Phone (any version) or iaxComm soft phone >> >> Please let me know if you've seen this before or if you can reproduce >> it. >> >> Thanks, >> >> Steve >> >> P.S. - A multi-codec version of IAX Phone will be out soon. >> > > > > ------------------------------------------------------- > This SF.Net email is sponsored by: > Sybase ASE Linux Express Edition - download now for FREE > LinuxWorld Reader's Choice Award Winner for best database on Linux. > http://ads.osdn.com/?ad_id=5588&alloc_id=12065&op=click > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
From: Adam H. <ad...@te...> - 2004-11-08 04:24:01
|
Both ends? any differences in ethereal? jitterbuffer off? Steven Sokol wrote: > I seem to have hit a problem with calls between users on IAX Phone and > the popular (and expensive) Cisco 7960. The audio quality is TERRIBLE. > Broken up, ragged sounding. It makes no difference which codecs I use > on either end (this is an experimental version of IAX Phone with > multiple codecs). The same problem seems to occur with iaxComm and > Firefly as well. > > It ONLY appears to happen with the Cisco. My other SIP phones (a > Grandstream BT101, a Uniden, an ACT, a Sipura and an ariaVoice) all work > just fine with the incoming audio stream from an IAX Client device. > > It also appears to be a problem with iaxClient calls, as IAX2 calls > Asterisk-to-Asterisk sound fine. > > Can anybody else out there with a Cisco 7960 set up and test this? > Here's my current configuration: > > Cisco 7960 w/ SIP Image 7.1.00 > Asterisk Stable-1.0.1 > IAX Phone (any version) or iaxComm soft phone > > Please let me know if you've seen this before or if you can reproduce it. > > Thanks, > > Steve > > P.S. - A multi-codec version of IAX Phone will be out soon. > |
From: Steven S. <ss...@so...> - 2004-11-08 03:59:50
|
I seem to have hit a problem with calls between users on IAX Phone and the popular (and expensive) Cisco 7960. The audio quality is TERRIBLE. Broken up, ragged sounding. It makes no difference which codecs I use on either end (this is an experimental version of IAX Phone with multiple codecs). The same problem seems to occur with iaxComm and Firefly as well. It ONLY appears to happen with the Cisco. My other SIP phones (a Grandstream BT101, a Uniden, an ACT, a Sipura and an ariaVoice) all work just fine with the incoming audio stream from an IAX Client device. It also appears to be a problem with iaxClient calls, as IAX2 calls Asterisk-to-Asterisk sound fine. Can anybody else out there with a Cisco 7960 set up and test this? Here's my current configuration: Cisco 7960 w/ SIP Image 7.1.00 Asterisk Stable-1.0.1 IAX Phone (any version) or iaxComm soft phone Please let me know if you've seen this before or if you can reproduce it. Thanks, Steve P.S. - A multi-codec version of IAX Phone will be out soon. |
From: Steve K. <st...@st...> - 2004-11-08 02:46:04
|
On Nov 7, 2004, at 12:02 PM, Dan wrote: > Hi, > > There is someone else having problems with the uLaw codec in iaxclient > livbrary? > This is the scenario for me: > - ATA186 SIP, uLaw preferred codec, uLaw and aLaw only supported > - DIAX with the latest iaxclient library, iLBC preferred codec, > uLaw/GSM/Speex supported. > > Calling from DIAX to ATA, iLBC codec is used, Asterisk make the > conversion. Ths ound is ok in both directions. > Calling from ATA to DIAX, uLaw is selected. > The sound from DIAX to ATA is perfect. > The sound from ATA to DIAX is very choppy and with very high latency > (several seconds). > > Some thoughs about that? There should be four different audio situations, but you're only reporting the result in two cases: Call from iaxclient to ATA, with iLBC on the iaxclient side, and asterisk translating: Case 1: Audio from iaxclient to ATA. Case 2: Audio from ATA to iaxclient. Call from ATA to iaxcleint, uLaw all the way Case 3: Audio from iaxclient to ATA. Case 4: Audio from ATA to iaxclient. Which audio cases result in problems? In those cases, you should take a look at the network traffic via ethereal, and see if the times and timestamps being sent into iaxclient make sense. For uLaw, you should get frames about every 20ms, and the timestamps should be separated by 20ms. For iLBC, it should be 30ms with the way asterisk works. -SteveK |
From: Steve K. <st...@st...> - 2004-11-08 02:38:34
|
On Nov 7, 2004, at 8:27 PM, Michael Van Donselaar wrote: > <snip> > >>> The same for the current used codec. >> >> I'm not sure what you mean here. I won't make iLBC the default, >> because it is not free. >> >> Both of these things could be made options configurable from the >> Makefile, though. > > We need to add IAX_FORMAT_ILBC to audio_format_capability in > iaxclient_lib.c, if we have ILBC defined in the Makefile. I can post > the > change to CVS. OK > Would it be acceptable if I also did: > > #ifdef CODEC_ILBC > audio_format_preferred = IAXC_FORMAT_ILBC > #else > audio_format_preferred = IAXC_FORMAT_SPEEX > #endif > > I'm suggesting that if ILBC is compiled in, it's also preferred. Personally, I prefer speex, but I suspect that I'd lose in a vote. Maybe after I add the code to select and control the VBR modes from speex, I'd be able to change minds. So, while it's pretty trivial for people to change this in their implementations, and I'd prefer to encourage people to use and contribute to speex, I won't veto this. -SteveK |
From: Michael V. D. <mi...@va...> - 2004-11-08 01:27:51
|
<snip> > > The same for the current used codec. > > I'm not sure what you mean here. I won't make iLBC the default, > because it is not free. > > Both of these things could be made options configurable from the > Makefile, though. We need to add IAX_FORMAT_ILBC to audio_format_capability in iaxclient_lib.c, if we have ILBC defined in the Makefile. I can post the change to CVS. Would it be acceptable if I also did: #ifdef CODEC_ILBC audio_format_preferred = IAXC_FORMAT_ILBC #else audio_format_preferred = IAXC_FORMAT_SPEEX #endif I'm suggesting that if ILBC is compiled in, it's also preferred. > > -SteveK --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004 |
From: Steve K. <st...@st...> - 2004-11-08 00:30:22
|
On Nov 6, 2004, at 6:49 AM, Sanjeev Saxena wrote: > Hi All, > > Recently I have started developing an application > using IAXClient library. > I am using iaxc_register() to register my application > with asterisk server. > > Could anyone tell me, what is the procedure for > unregistering my client app ? There isn't one yet. Actually, I don't think IAX2 has an unregister command, but one could modify iaxclient to keep track of registrations, and stop re-registering. I think this was discussed on the list recently. (or, it might have been in a private e-mail..). -SteveK |
From: Steve K. <st...@st...> - 2004-11-08 00:25:54
|
On Nov 7, 2004, at 2:49 PM, Dan wrote: > Hi, > >> ----- Original Message ----- From: "Babar Shafiq" >> <bab...@ya...> >> ... >> I m only testing it on windows (98,2000,XP), with my IaxClientOCX, I >> also noticed the dealy of one >> or two seconds with the half duplex behaviour of audio (when using >> ilbc, speex). How to minimize >> this delay ? and increase volume of audio ? (little low after Analog >> AGC patch). >> > > Try to set the environment variable (even from inside your app, like > in DIAX): > PA_MIN_LATENCY_MSEC > A value of 60 seems to be the best for me (I think that the default is > 400). > > Steve, what about including this in the iaxclient library, as default? I could do that; I guess I haven't yet been convinced that this isn't going to cause problems when machines are heavily loaded. > The same for the current used codec. I'm not sure what you mean here. I won't make iLBC the default, because it is not free. Both of these things could be made options configurable from the Makefile, though. -SteveK |
From: Dan <dt...@fx...> - 2004-11-07 19:50:00
|
Hi, >----- Original Message ----- >From: "Babar Shafiq" <bab...@ya...> > ... > I m only testing it on windows (98,2000,XP), with my IaxClientOCX, I also > noticed the dealy of one > or two seconds with the half duplex behaviour of audio (when using ilbc, > speex). How to minimize > this delay ? and increase volume of audio ? (little low after Analog AGC > patch). > Try to set the environment variable (even from inside your app, like in DIAX): PA_MIN_LATENCY_MSEC A value of 60 seems to be the best for me (I think that the default is 400). Steve, what about including this in the iaxclient library, as default? The same for the current used codec. I think is better to be included in the library and not in each client separately Best regards, Dan |
From: Babar S. <bab...@ya...> - 2004-11-07 19:06:06
|
Hi List and Steve, I m only testing it on windows (98,2000,XP), with my IaxClientOCX, I also noticed the dealy of one or two seconds with the half duplex behaviour of audio (when using ilbc, speex). How to minimize this delay ? and increase volume of audio ? (little low after Analog AGC patch). Regards, Babar Shafiq Nazmi. IaxClientOCX with current CVS updated http://203.170.71.26/IaxClientSetup.exe, http://www.geocities.com/babarnazmi/iaxclient-snap.jpg > Date: Sun, 07 Nov 2004 01:27:19 +0100 > From: daniel huhardeaux <de...@to...> > To: iax...@li... > Subject: Re: [Iaxclient-devel] Next to Audio Caller problem > > Michael Van Donselaar a =E9crit : > > >On Sat, 06 Nov 2004 19:11:26 +0100, daniel huhardeaux <de...@to...>= > wrote: > > > > =20 > > > >>Hi, > >> > >>having still my problem with iaxcomm under linux, self compiled version= > =20 > >>or last binaries from sf, I decide to test under w2k WS. I had Diax097a= > ,=20 > >>iaxcomm CVS Februar 04 and Iaxphone 0.1.0. This machine is connected to= > =20 > >>the same asterisk server that my notebook is using: everything is fine,= > =20 > >>all calls are clear both sides. > >> > >>Now I decide to install latest version of this software, taken from=20 > >>sf.net Diax098c, iaxcomm CVS October 04 and Iaxphone 0.2.0 build 116 > >> > >>Launching: > >> > >>diax098c: class not registered - MSSTDMFT.DLL has to be installed on=20 > >>your machine > >> =20 > >> > > > >Did you ever get it to run successfully? > > =20 > > > Diax097a yes. For Diax 098c, even after having installed this dll in=20 > WINNT/system or WINNT/system32 I get this message. > > > =20 > > > >>iaxcomm: I can call asterisk demo on my server, no problem. I try to=20 > >>call echo test from FWD, segfault. In w2k event logs I see " ...=20 > >>exception was c0000005 at address 0041CDD5 9<no symbols>) FWD is using=20 > >>only ULAW, so I guess problem come from here. > >> =20 > >> > > > >Which version of iaxComm are these results for? > > =20 > > > Windows zip from october 2004 taken from SF. > > > =20 > > > >>iaxphone: start with errors initializing the DirectSound drivers > >> > >>Error numbers=3D0 Error description =3D > >> > >>and a Note about ound card not installed or in use. Clicking OK appears= > =20 > >>same error, clicking OK a window telling me that it was an error with=20 > >>DirectX and DirectMusic modules. I pass clicking OK I call again FWD=20 > >>echo test and it's ok. > >> =20 > >> > > > > =20 > > > >>Now I compile tkphone and test it: I get my audio but with a delay of=20 > >>more then 6s (under w2k if there was delay, maxi 1s) I immediately stop= > =20 > >>the call, launch iasxcomm: as usually. I restart tkphone, same result a= > s=20 > >>the first time. So definitely problem is with iaxcomm. > >> =20 > >> > > > >I'm not sure I understand this last paragraph. It's probably better to = > be > >verbose than concise when describing a problem. > > =20 > > > Sorry. All this paragraph is concerning linux > > >I'm getting that you tested tkphone both on linux and on Win2k WS. tkph= > one on > >win2k gives a 1 sec audio delay; tkphone on linux gives 6 sec audio dela= > y. > > > >After stopping tkphone on one of the platforms (which one?), > > > Only linux > > > you immediately run > >some version of iaxComm (which one?). > > > linux from october 2004 taken from SF or self compiled > > > iaxComm then gives you the usual problem > >when you call (is this the segfault, or ugly audio?). > > > ugly audio > > > Then you restart tkphone > >and get the same result (I assume 1-6 second audio delay). > > =20 > > > Yes, audio with delay. > > >While the problem really may be with the iaxComm code, I don't agree tha= > t you've > >isolated it. > > > I don't say I isolate problem. Being the only one to have such=20 > behaviour, it would be pretentious. I'm trying to find some trail. > > > I know that DIAX and IAXPhone are not using CVS code. > > > >IAXPhone is using iaxclient code from before any other codecs were added= > . You > >mention version 0.97a and 0.98c of DIAX, but don't show a successful run. > > > 0.97a is successful > > > Even > >a 1 second audio delay for tkphone sounds unsuccessful to me. > > > >What results do you get on Win2k when running > >http://iaxclient.sourceforge.net/iaxcomm/iaxcomm-win-20040228.zip ? > > =20 > > > As sad at the begining of my mail: is working. To summarize, taken only=20 > audio problem in account: > > Diax097a, iaxcomm 02/2004, IaxPhone 0.1.0 and 0.2.0 (all windows),=20 > tkphone (linux) are working. > > Thanks for your help > > --=20 > Daniel> ===== God is a great Programmer __________________________________ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com |
From: Michael V. D. <mi...@va...> - 2004-11-07 18:27:54
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> -----Original Message----- > From: iax...@li...=20 > [mailto:iax...@li...] On=20 > Behalf Of daniel huhardeaux > Sent: Saturday, November 06, 2004 11:41 PM > To: iaxclient-list > Subject: Re: [Iaxclient-devel] Next to Audio Caller problem >=20 > Michael Van Donselaar a =E9crit : >=20 > >>As sad at the begining of my mail: is working. To summarize, taken=20 > >>only audio problem in account: > >> > >>Diax097a, iaxcomm 02/2004, IaxPhone 0.1.0 and 0.2.0 (all windows),=20 > >>tkphone (linux) are working. > >> =20 > >> > > > >There are only two changes to the iaxcomm code between Feb=20 > 04 and Oct 04: > > > >1) Feb code had a null pointer function reference when right=20 > clicking=20 > >on the tray icon. This could not affect linux port, as it is in a=20 > >block that is ifdefed out > > > >2) Clicking Dial now clears the extension combobox. > > > >Any other changes in behavior you are seeing are not iaxcomm=20 > specific. > > =20 > > > Two questions: >=20 > 1. why iaxcomm windows from october segfault? I strongly suspect that it has something to do with the codec = negotiation code. Other than the two changes noted above, there have been no = changes to iaxcomm, only (many!) changes to iaxclient. I see in another message from Dan, the author of DIAX, that he is = getting complaints about ulaw audio quality. I also noticed that ulaw was = mentioned in on of the error messages you got. Since you are compiling yourself already, and I think I recall you = prefer iLBC, try this: Add=20 iaxc_set_formats(IAXC_FORMAT_ILBC, IAXC_FORMAT_ILBC); right after=20 iaxc_start_processing_thread(); in main.cc (approx line 206). This will force an iLBC connection. = Since you have hardware for both platforms, I'd be interested to see how this works for you. > 2. I test again with iaxcomm linux from februar: I have audio=20 > with distortion But did I understand correctly that you got good results with iaxcomm-win-20040228.zip? > What I also notice, is that from time to time iaxcomm is not=20 > exiting properly. Closing the main window has no action.=20 > Going to terminal from where iaxcomm was launcheg and ctrl/c.=20 > Nextime iaxcomm is started, the lock file is deleted. >=20 > I suggested some time ago to call someone from you so you=20 > will be able to hear what happends. Still actual ;-) And as I responded, I do not have any hardware to make this practical. = I have only one linux box with a sound card, and it is in the basement = without any speaker or microphone. >=20 > -- > Daniel >=20 >=20 > ------------------------------------------------------- > This SF.Net email is sponsored by: > Sybase ASE Linux Express Edition - download now for FREE > LinuxWorld Reader's Choice Award Winner for best database on Linux. > http://ads.osdn.com/?ad_idU88&alloc_id=12065&op=3Dclick > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel >=20 > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004 > =20 >=20 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 11/1/2004 =20 |
From: Ilguiz L. <ila...@in...> - 2004-11-07 17:52:14
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On Sun, Nov 07, 2004 at 12:51:06PM -0500, Ilguiz Latypov wrote: > The incoming voice data are probably never sent to /dev/dsp by > software. I meant microphone data (outgoing network wise). -- Ilguiz |
From: Ilguiz L. <ila...@in...> - 2004-11-07 17:50:19
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On Sun, Nov 07, 2004 at 07:02:45PM +0200, Dan wrote: > The sound from ATA to DIAX is very choppy and with very high latency > (several seconds). I am mostly calling from iaxphone via an IAX2 provider and experience crackle at times. I once tried to record the sound along with the network exchange. I used vsound with its LD_PRELOAD trick. The recorded incoming audio was perfect even when I heard crackle over the speakers at the same time. (I was confused about my outgoing audio not recorded by vsound. Now I realized this should have been expected because my own voice I heard from the speakers might only be a result of a hardware loop. The incoming voice data are probably never sent to /dev/dsp by software.) So I am suspecting that the crackle might be due to the portaudio/OSS implementation. Unfortunately, the newer v.19 portaudio with the ALSA backend doesn't seem to have a mixer interface yet. Regards, -- Ilguiz |
From: Dan <dt...@fx...> - 2004-11-07 17:02:54
|
Hi, There is someone else having problems with the uLaw codec in iaxclient livbrary? This is the scenario for me: - ATA186 SIP, uLaw preferred codec, uLaw and aLaw only supported - DIAX with the latest iaxclient library, iLBC preferred codec, uLaw/GSM/Speex supported. Calling from DIAX to ATA, iLBC codec is used, Asterisk make the conversion. Ths ound is ok in both directions. Calling from ATA to DIAX, uLaw is selected. The sound from DIAX to ATA is perfect. The sound from ATA to DIAX is very choppy and with very high latency (several seconds). Some thoughs about that? Thank you, Dan |
From: Dan <dt...@fx...> - 2004-11-07 15:59:43
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>----- Original Message ----- >From: "daniel huhardeaux" <de...@to...> > > Installed, launched and working. Perfect audio, no delay, everything is > fine. DIAX 0.9.9e will be better than that and we'll support an USB phone too (automatically detected and configured) Best regards, Dan |
From: daniel h. <dan...@to...> - 2004-11-07 15:03:30
|
Michael Van Donselaar a =E9crit : >>As sad at the begining of my mail: is working. To summarize,=20 >>taken only audio problem in account: >> >>Diax097a, iaxcomm 02/2004, IaxPhone 0.1.0 and 0.2.0 (all=20 >>windows), tkphone (linux) are working. >> =20 >> > >There are only two changes to the iaxcomm code between Feb 04 and Oct 04= : > >1) Feb code had a null pointer function reference when right clicking on= the >tray icon. This could not affect linux port, as it is in a block that i= s >ifdefed out > >2) Clicking Dial now clears the extension combobox. > >Any other changes in behavior you are seeing are not iaxcomm specific. > =20 > Two questions: 1. why iaxcomm windows from october segfault? 2. I test again with iaxcomm linux from februar: I have audio with=20 distortion What I also notice, is that from time to time iaxcomm is not exiting=20 properly. Closing the main window has no action. Going to terminal from=20 where iaxcomm was launcheg and ctrl/c. Nextime iaxcomm is started, the=20 lock file is deleted. I suggested some time ago to call someone from you so you will be able=20 to hear what happends. Still actual ;-) --=20 Daniel |
From: daniel h. <de...@to...> - 2004-11-07 11:25:17
|
Dan a =E9crit : > > Hi Daniel, Hi Dan > >>> Did you ever get it to run successfully? >>> >> Diax097a yes. For Diax 098c, even after having installed this dll in >> WINNT/system or WINNT/system32 I get this message. >> > This is a bug in 0.9.8c which appear when WinXP SP2 is installed. > Try to use the new prerelease of 0.9.9.x from: > http://www.geocities.com/tdanro/diax/diax099b.zip Installed, launched and working. Perfect audio, no delay, everything is=20 fine. --=20 Daniel |
From: daniel h. <de...@to...> - 2004-11-07 10:37:18
|
Dan a =E9crit : > > Hi Daniel, > >>> Did you ever get it to run successfully? >>> >> Diax097a yes. For Diax 098c, even after having installed this dll in >> WINNT/system or WINNT/system32 I get this message. >> > This is a bug in 0.9.8c which appear when WinXP SP2 is installed. It's W2k not XP. > > Try to use the new prerelease of 0.9.9.x from: > http://www.geocities.com/tdanro/diax/diax099b.zip > to solve this issue or reregister the DLL with regsrv32 . Ok, will try [...] > DIAX 0.9.9b use a verion of Iaxclient from allmost one month ago. > The updated version will be 0.9.9e So will be more next my setup. Will give it a try > > Pls send me your feedback. Sure ;-) --=20 Daniel |
From: Dan <dt...@fx...> - 2004-11-07 07:51:01
|
Hi Daniel, >>Did you ever get it to run successfully? >> > Diax097a yes. For Diax 098c, even after having installed this dll in > WINNT/system or WINNT/system32 I get this message. > This is a bug in 0.9.8c which appear when WinXP SP2 is installed. Try to use the new prerelease of 0.9.9.x from: http://www.geocities.com/tdanro/diax/diax099b.zip to solve this issue or reregister the DLL with regsrv32 . DIAX 0.9.9e will be available during the next week. > Yes, audio with delay. In DIAX 0.9.9x you have an option for Audio Latency setting >> I know that DIAX and IAXPhone are not using CVS code. DIAX 0.9.9b use a verion of Iaxclient from allmost one month ago. The updated version will be 0.9.9e >> >>IAXPhone is using iaxclient code from before any other codecs were added. >>You >>mention version 0.97a and 0.98c of DIAX, but don't show a successful run. >> > 0.97a is successful See my first comment. > Diax097a, iaxcomm 02/2004, IaxPhone 0.1.0 and 0.2.0 (all windows), tkphone > (linux) are working. > Pls send me your feedback. Best regards, Dan _______________________________________________________________ Connex scaneaza automat toate mesajele impotriva virusilor folosind Trend Micro VirusWall. Connex automatically scans all messages for viruses using Trend Micro VirusWall. _______________________________________________________________ Nota: Este posibil ca produsul Trend Micro VirusWall sa nu detecteze toti virusii noi sau toate variantele lor. Va rugam sa luati in considerare ca exista un risc de fiecare data cind deschideti fisiere atasate si ca MobiFon nu este responsabila pentru orice prejudiciu cauzat de decizia dvs. Disclaimer: It is possible that the Trend Micro VirusWall product may not be able to detect all new viruses and variants. Please be aware that there is a risk involved whenever opening e-mail attachments to your computer and that MobiFon is not responsible for any damages caused by your decision to do so. |