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From: Adam H. <ad...@te...> - 2004-02-05 00:41:38
|
Why don't you guys just use my code from firefly, it's has nanosecond
accuracy. Beats the hell out of 10ms accuaracy
static __int64 freq,start;
static int inited = 0;
//static time_t startuptime;
BOOL APIENTRY DllMain( HANDLE hModule,
DWORD ul_reason_for_call,
LPVOID lpReserved
)
{
if(!inited)
{
inited = 1;
QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
QueryPerformanceCounter((LARGE_INTEGER*)&start);
}
return TRUE;
}
void gettimeofday(struct timeval *tv, struct timezone *tz)
{
__int64 time;
double elapsed;
QueryPerformanceCounter((LARGE_INTEGER*)&time);
elapsed = (double)(time - start) / (double)freq;
tv->tv_sec = (long)elapsed;
tv->tv_usec = (long)((elapsed-tv->tv_sec) * 1000000);
}
enjoy,
Adam
----- Original Message -----
From: "Steven Sokol" <ss...@so...>
To: <iax...@li...>
Sent: Thursday, February 05, 2004 11:34 AM
Subject: RE: [Iaxclient-devel] Some data related to the new bug...
> Yes. Guilty as charged. It always happens UNDER WINDOWS.
>
> I believe that the gettimeofday() is being replaced by a function based on
> GetTickCount(). I wonder if that doesn't have something to do with it.
>
> Steve S
>
> -----Original Message-----
> From: iax...@li...
> [mailto:iax...@li...] On Behalf Of Steve
> Underwood
> Sent: Wednesday, February 04, 2004 5:48 PM
> To: iax...@li...
> Subject: Re: [Iaxclient-devel] Some data related to the new bug...
>
> Steven Sokol wrote:
>
> >More notes:
> >
> >1. It ALWAYS happens. It is not an intermittent issue. This happens
for
> >EVERY IAX2-to-IAX2 call made using iaxClient.
> >
> >
> Not EVERY call. I make hour long calls using iaxclient IAX2-to-IAX2 and
> have no trouble. I use Linux. Is that the difference?
>
> >2. The timing is uncanny. It ALWAYS happens after 65-67 seconds.
> >
> >3. The audio dies but the call in not immediately torn down. The call
> >eventually is eventually killed when the PING/PONG cycle times out some
> >seconds later (perhaps over a minute in some of my tests).
> >
> >4. In EVERY case, the Asterisk kicks out the aforementioned voice frame
> >retransmit message.
> >
> >
> As Adam said, there is something wrong if a voice frame is being
> retransmitted at that point in the call. Perhaps checking places where
> the frame is tagged for retransmission would home in on the problem.
>
> >5. I don't see registration messages coming through at the same time. I
> >was originally guessing that this happened during the re-reg process.
Not
> >so (or at least that doesn't seem to be the case -- could be that the
> re-reg
> >that takes place prior to the error causes a problem).
> >
> >
> Regards,
> Steve
>
>
>
>
> -------------------------------------------------------
> The SF.Net email is sponsored by EclipseCon 2004
> Premiere Conference on Open Tools Development and Integration
> See the breadth of Eclipse activity. February 3-5 in Anaheim, CA.
> http://www.eclipsecon.org/osdn
> _______________________________________________
> Iaxclient-devel mailing list
> Iax...@li...
> https://lists.sourceforge.net/lists/listinfo/iaxclient-devel
>
>
>
>
> -------------------------------------------------------
> The SF.Net email is sponsored by EclipseCon 2004
> Premiere Conference on Open Tools Development and Integration
> See the breadth of Eclipse activity. February 3-5 in Anaheim, CA.
> http://www.eclipsecon.org/osdn
> _______________________________________________
> Iaxclient-devel mailing list
> Iax...@li...
> https://lists.sourceforge.net/lists/listinfo/iaxclient-devel
|
|
From: Steven S. <ss...@so...> - 2004-02-05 00:34:39
|
Yes. Guilty as charged. It always happens UNDER WINDOWS. I believe that the gettimeofday() is being replaced by a function based on GetTickCount(). I wonder if that doesn't have something to do with it. Steve S -----Original Message----- From: iax...@li... [mailto:iax...@li...] On Behalf Of Steve Underwood Sent: Wednesday, February 04, 2004 5:48 PM To: iax...@li... Subject: Re: [Iaxclient-devel] Some data related to the new bug... Steven Sokol wrote: >More notes: > >1. It ALWAYS happens. It is not an intermittent issue. This happens for >EVERY IAX2-to-IAX2 call made using iaxClient. > > Not EVERY call. I make hour long calls using iaxclient IAX2-to-IAX2 and have no trouble. I use Linux. Is that the difference? >2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. > >3. The audio dies but the call in not immediately torn down. The call >eventually is eventually killed when the PING/PONG cycle times out some >seconds later (perhaps over a minute in some of my tests). > >4. In EVERY case, the Asterisk kicks out the aforementioned voice frame >retransmit message. > > As Adam said, there is something wrong if a voice frame is being retransmitted at that point in the call. Perhaps checking places where the frame is tagged for retransmission would home in on the problem. >5. I don't see registration messages coming through at the same time. I >was originally guessing that this happened during the re-reg process. Not >so (or at least that doesn't seem to be the case -- could be that the re-reg >that takes place prior to the error causes a problem). > > Regards, Steve ------------------------------------------------------- The SF.Net email is sponsored by EclipseCon 2004 Premiere Conference on Open Tools Development and Integration See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. http://www.eclipsecon.org/osdn _______________________________________________ Iaxclient-devel mailing list Iax...@li... https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
|
From: Kim H. <ki...@ki...> - 2004-02-05 00:15:55
|
> If this only happens on windows, one place I'd look it to make sure the > "gettimeofday" replacement there is correct; it's in winfuncs.c. It's > basically adapted from the earlier hack in libiax "winiphone".. It fails for me with IAXCLIENT -> asterisk type of calls always as well. - Kim |
|
From: Steve K. <st...@st...> - 2004-02-04 23:59:28
|
Steve Underwood wrote: > Steven Sokol wrote: > >> More notes: >> >> 1. It ALWAYS happens. It is not an intermittent issue. This >> happens for >> EVERY IAX2-to-IAX2 call made using iaxClient. >> >> > Not EVERY call. I make hour long calls using iaxclient IAX2-to-IAX2 > and have no trouble. I use Linux. Is that the difference? Glad to hear it is working well for you, though :) >> 2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. >> >> 3. The audio dies but the call in not immediately torn down. The call >> eventually is eventually killed when the PING/PONG cycle times out some >> seconds later (perhaps over a minute in some of my tests). >> >> 4. In EVERY case, the Asterisk kicks out the aforementioned voice frame >> retransmit message. >> >> > As Adam said, there is something wrong if a voice frame is being > retransmitted at that point in the call. Perhaps checking places where > the frame is tagged for retransmission would home in on the problem. If you look at the code I excerpted, it looks like voice frames are occasionally sent as full frames. I'm not sure of the logic behind that (I'd say it would be good if we didn't have all that Ping and Lagrq stuff already being sent, in order to refresh timeouts and establish connectivity with acks). It also correlates exactly with his 65-67 second timing. If this only happens on windows, one place I'd look it to make sure the "gettimeofday" replacement there is correct; it's in winfuncs.c. It's basically adapted from the earlier hack in libiax "winiphone".. |
|
From: Steve K. <st...@st...> - 2004-02-04 23:53:52
|
Steven Sokol wrote: >I don't know about Firefly (yet). I will have to check that out. > >Does anybody know what could cause the clients (both ends) to loose their >audio streams? There are two streams here - the inbound and the outbound. >They both pass through Asterisk. > >I guess in reality there are three parties to the call: > [USER A]<->[ASTERISK]<->[USER B] > >So there are actually 4 streams. Something is killing off all of them. >Would this be happening in Asterisk? How do we trace this on the client >side? > >Steve K, any thoughts? Can we do anything to capture additional information >or to tell of the audio streams are dying locally or at Asterisk? > > You can use tcpdump/ethereal on the asterisk server box to capture and examine the actual packets that are sent in each direction. You can turn on debugging, so that chan_iax2 prints something whenever a packet is sent, doing something similar. Are these calls getting "natively" bridged somehow? [so the frames get sent directly from user a to user b?) I don't know if the library supports that or not; Maybe kram has some idea (he is on this list now..). -SteveK >Thanks, > >Steve > > > > > >>-----Original Message----- >>From: iax...@li... [mailto:iaxclient-devel- >>ad...@li...] On Behalf Of Adam Hart >>Sent: Wednesday, February 04, 2004 5:02 PM >>To: iax...@li... >>Subject: Re: [Iaxclient-devel] Some data related to the new bug... >> >>I'm not trying to plug firefly, but does the same issue occur on it? >> >>----- Original Message ----- >>From: "Steven Sokol" <ss...@so...> >>To: <iax...@li...> >>Sent: Thursday, February 05, 2004 10:01 AM >>Subject: RE: [Iaxclient-devel] Some data related to the new bug... >> >> >> >> >>>More notes: >>> >>>1. It ALWAYS happens. It is not an intermittent issue. This happens >>> >>> >>for >> >> >>>EVERY IAX2-to-IAX2 call made using iaxClient. >>> >>>2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. >>> >>>3. The audio dies but the call in not immediately torn down. The call >>>eventually is eventually killed when the PING/PONG cycle times out some >>>seconds later (perhaps over a minute in some of my tests). >>> >>>4. In EVERY case, the Asterisk kicks out the aforementioned voice frame >>>retransmit message. >>> >>>5. I don't see registration messages coming through at the same time. >>> >>> >>I >> >> >>>was originally guessing that this happened during the re-reg process. >>> >>> >>Not >> >> >>>so (or at least that doesn't seem to be the case -- could be that the >>> >>> >>re-reg >> >> >>>that takes place prior to the error causes a problem). >>> >>> >>> >>>>-----Original Message----- >>>>From: iax...@li... >>>> >>>> >>[mailto:iaxclient-devel- >> >> >>>>ad...@li...] On Behalf Of Steven Sokol >>>>Sent: Wednesday, February 04, 2004 4:44 PM >>>>To: iax...@li... >>>>Subject: [Iaxclient-devel] Some data related to the new bug... >>>> >>>>1. I cannot make the bug reproduce between my SIP client and my IAX >>>> >>>> >>Phone >> >> >>>>instance. It simply does not happen. It also does not happen when >>>> >>>> >>calls >> >> >>>>are placed between IAX Phone and various asterisk service (i.e. >>>> >>>> >>Voicemail, >> >> >>>>MeetMe, MOH, etc.) >>>> >>>>2. The audio seems to die at the point when Asterisk transmits a >>>> >>>> >>retry >> >> >>>>packet: >>>> >>>>Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: >>>> >>>> >>2 >> >> >>>> Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] >>>>sip*CLI> >>>>Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: >>>> >>>> >>2 >> >> >>>> Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] >>>> >>>>The CLI> above indicates where I hit the enter key just as soon as I >>>> >>>> >>heard >> >> >>>>the audio die off. >>>> >>>>It seems the issue is specific to IAX2 and possibly to the iaxClient >>>>library. >>>> >>>>Does this ring any bells for anyone? >>>> >>>>Thanks, >>>> >>>>Steve >>>> >>>> >>>> >>>> >>>>------------------------------------------------------- >>>>The SF.Net email is sponsored by EclipseCon 2004 >>>>Premiere Conference on Open Tools Development and Integration >>>>See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. >>>>http://www.eclipsecon.org/osdn >>>>_______________________________________________ >>>>Iaxclient-devel mailing list >>>>Iax...@li... >>>>https://lists.sourceforge.net/lists/listinfo/iaxclient-devel >>>> >>>> >>> >>> >>>------------------------------------------------------- >>>The SF.Net email is sponsored by EclipseCon 2004 >>>Premiere Conference on Open Tools Development and Integration >>>See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. >>>http://www.eclipsecon.org/osdn >>>_______________________________________________ >>>Iaxclient-devel mailing list >>>Iax...@li... >>>https://lists.sourceforge.net/lists/listinfo/iaxclient-devel >>> >>> >> >>------------------------------------------------------- >>The SF.Net email is sponsored by EclipseCon 2004 >>Premiere Conference on Open Tools Development and Integration >>See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. >>http://www.eclipsecon.org/osdn >>_______________________________________________ >>Iaxclient-devel mailing list >>Iax...@li... >>https://lists.sourceforge.net/lists/listinfo/iaxclient-devel >> >> > > > > >------------------------------------------------------- >The SF.Net email is sponsored by EclipseCon 2004 >Premiere Conference on Open Tools Development and Integration >See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. >http://www.eclipsecon.org/osdn >_______________________________________________ >Iaxclient-devel mailing list >Iax...@li... >https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > > |
|
From: Steve U. <st...@co...> - 2004-02-04 23:48:09
|
Steven Sokol wrote: >More notes: > >1. It ALWAYS happens. It is not an intermittent issue. This happens for >EVERY IAX2-to-IAX2 call made using iaxClient. > > Not EVERY call. I make hour long calls using iaxclient IAX2-to-IAX2 and have no trouble. I use Linux. Is that the difference? >2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. > >3. The audio dies but the call in not immediately torn down. The call >eventually is eventually killed when the PING/PONG cycle times out some >seconds later (perhaps over a minute in some of my tests). > >4. In EVERY case, the Asterisk kicks out the aforementioned voice frame >retransmit message. > > As Adam said, there is something wrong if a voice frame is being retransmitted at that point in the call. Perhaps checking places where the frame is tagged for retransmission would home in on the problem. >5. I don't see registration messages coming through at the same time. I >was originally guessing that this happened during the re-reg process. Not >so (or at least that doesn't seem to be the case -- could be that the re-reg >that takes place prior to the error causes a problem). > > Regards, Steve |
|
From: Steven S. <ss...@so...> - 2004-02-04 23:35:53
|
I don't know about Firefly (yet). I will have to check that out. Does anybody know what could cause the clients (both ends) to loose their audio streams? There are two streams here - the inbound and the outbound. They both pass through Asterisk. I guess in reality there are three parties to the call: [USER A]<->[ASTERISK]<->[USER B] So there are actually 4 streams. Something is killing off all of them. Would this be happening in Asterisk? How do we trace this on the client side? Steve K, any thoughts? Can we do anything to capture additional information or to tell of the audio streams are dying locally or at Asterisk? Thanks, Steve > -----Original Message----- > From: iax...@li... [mailto:iaxclient-devel- > ad...@li...] On Behalf Of Adam Hart > Sent: Wednesday, February 04, 2004 5:02 PM > To: iax...@li... > Subject: Re: [Iaxclient-devel] Some data related to the new bug... > > I'm not trying to plug firefly, but does the same issue occur on it? > > ----- Original Message ----- > From: "Steven Sokol" <ss...@so...> > To: <iax...@li...> > Sent: Thursday, February 05, 2004 10:01 AM > Subject: RE: [Iaxclient-devel] Some data related to the new bug... > > > > More notes: > > > > 1. It ALWAYS happens. It is not an intermittent issue. This happens > for > > EVERY IAX2-to-IAX2 call made using iaxClient. > > > > 2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. > > > > 3. The audio dies but the call in not immediately torn down. The call > > eventually is eventually killed when the PING/PONG cycle times out some > > seconds later (perhaps over a minute in some of my tests). > > > > 4. In EVERY case, the Asterisk kicks out the aforementioned voice frame > > retransmit message. > > > > 5. I don't see registration messages coming through at the same time. > I > > was originally guessing that this happened during the re-reg process. > Not > > so (or at least that doesn't seem to be the case -- could be that the > re-reg > > that takes place prior to the error causes a problem). > > > > > -----Original Message----- > > > From: iax...@li... > [mailto:iaxclient-devel- > > > ad...@li...] On Behalf Of Steven Sokol > > > Sent: Wednesday, February 04, 2004 4:44 PM > > > To: iax...@li... > > > Subject: [Iaxclient-devel] Some data related to the new bug... > > > > > > 1. I cannot make the bug reproduce between my SIP client and my IAX > Phone > > > instance. It simply does not happen. It also does not happen when > calls > > > are placed between IAX Phone and various asterisk service (i.e. > Voicemail, > > > MeetMe, MOH, etc.) > > > > > > 2. The audio seems to die at the point when Asterisk transmits a > retry > > > packet: > > > > > > Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: > 2 > > > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > > > sip*CLI> > > > Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: > 2 > > > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > > > > > > The CLI> above indicates where I hit the enter key just as soon as I > heard > > > the audio die off. > > > > > > It seems the issue is specific to IAX2 and possibly to the iaxClient > > > library. > > > > > > Does this ring any bells for anyone? > > > > > > Thanks, > > > > > > Steve > > > > > > > > > > > > > > > ------------------------------------------------------- > > > The SF.Net email is sponsored by EclipseCon 2004 > > > Premiere Conference on Open Tools Development and Integration > > > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > > > http://www.eclipsecon.org/osdn > > > _______________________________________________ > > > Iaxclient-devel mailing list > > > Iax...@li... > > > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > > > > > > > > > ------------------------------------------------------- > > The SF.Net email is sponsored by EclipseCon 2004 > > Premiere Conference on Open Tools Development and Integration > > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > > http://www.eclipsecon.org/osdn > > _______________________________________________ > > Iaxclient-devel mailing list > > Iax...@li... > > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > > > ------------------------------------------------------- > The SF.Net email is sponsored by EclipseCon 2004 > Premiere Conference on Open Tools Development and Integration > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > http://www.eclipsecon.org/osdn > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
|
From: Adam H. <ad...@te...> - 2004-02-04 23:05:12
|
I'm not trying to plug firefly, but does the same issue occur on it? ----- Original Message ----- From: "Steven Sokol" <ss...@so...> To: <iax...@li...> Sent: Thursday, February 05, 2004 10:01 AM Subject: RE: [Iaxclient-devel] Some data related to the new bug... > More notes: > > 1. It ALWAYS happens. It is not an intermittent issue. This happens for > EVERY IAX2-to-IAX2 call made using iaxClient. > > 2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. > > 3. The audio dies but the call in not immediately torn down. The call > eventually is eventually killed when the PING/PONG cycle times out some > seconds later (perhaps over a minute in some of my tests). > > 4. In EVERY case, the Asterisk kicks out the aforementioned voice frame > retransmit message. > > 5. I don't see registration messages coming through at the same time. I > was originally guessing that this happened during the re-reg process. Not > so (or at least that doesn't seem to be the case -- could be that the re-reg > that takes place prior to the error causes a problem). > > > -----Original Message----- > > From: iax...@li... [mailto:iaxclient-devel- > > ad...@li...] On Behalf Of Steven Sokol > > Sent: Wednesday, February 04, 2004 4:44 PM > > To: iax...@li... > > Subject: [Iaxclient-devel] Some data related to the new bug... > > > > 1. I cannot make the bug reproduce between my SIP client and my IAX Phone > > instance. It simply does not happen. It also does not happen when calls > > are placed between IAX Phone and various asterisk service (i.e. Voicemail, > > MeetMe, MOH, etc.) > > > > 2. The audio seems to die at the point when Asterisk transmits a retry > > packet: > > > > Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 > > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > > sip*CLI> > > Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 > > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > > > > The CLI> above indicates where I hit the enter key just as soon as I heard > > the audio die off. > > > > It seems the issue is specific to IAX2 and possibly to the iaxClient > > library. > > > > Does this ring any bells for anyone? > > > > Thanks, > > > > Steve > > > > > > > > > > ------------------------------------------------------- > > The SF.Net email is sponsored by EclipseCon 2004 > > Premiere Conference on Open Tools Development and Integration > > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > > http://www.eclipsecon.org/osdn > > _______________________________________________ > > Iaxclient-devel mailing list > > Iax...@li... > > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > > > > > ------------------------------------------------------- > The SF.Net email is sponsored by EclipseCon 2004 > Premiere Conference on Open Tools Development and Integration > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > http://www.eclipsecon.org/osdn > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
|
From: Steven S. <ss...@so...> - 2004-02-04 23:01:11
|
More notes: 1. It ALWAYS happens. It is not an intermittent issue. This happens for EVERY IAX2-to-IAX2 call made using iaxClient. 2. The timing is uncanny. It ALWAYS happens after 65-67 seconds. 3. The audio dies but the call in not immediately torn down. The call eventually is eventually killed when the PING/PONG cycle times out some seconds later (perhaps over a minute in some of my tests). 4. In EVERY case, the Asterisk kicks out the aforementioned voice frame retransmit message. 5. I don't see registration messages coming through at the same time. I was originally guessing that this happened during the re-reg process. Not so (or at least that doesn't seem to be the case -- could be that the re-reg that takes place prior to the error causes a problem). > -----Original Message----- > From: iax...@li... [mailto:iaxclient-devel- > ad...@li...] On Behalf Of Steven Sokol > Sent: Wednesday, February 04, 2004 4:44 PM > To: iax...@li... > Subject: [Iaxclient-devel] Some data related to the new bug... > > 1. I cannot make the bug reproduce between my SIP client and my IAX Phone > instance. It simply does not happen. It also does not happen when calls > are placed between IAX Phone and various asterisk service (i.e. Voicemail, > MeetMe, MOH, etc.) > > 2. The audio seems to die at the point when Asterisk transmits a retry > packet: > > Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > sip*CLI> > Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > > The CLI> above indicates where I hit the enter key just as soon as I heard > the audio die off. > > It seems the issue is specific to IAX2 and possibly to the iaxClient > library. > > Does this ring any bells for anyone? > > Thanks, > > Steve > > > > > ------------------------------------------------------- > The SF.Net email is sponsored by EclipseCon 2004 > Premiere Conference on Open Tools Development and Integration > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > http://www.eclipsecon.org/osdn > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
|
From: Adam H. <ad...@te...> - 2004-02-04 22:48:33
|
this may be a stupid question but why is asterisk retransmitting a voice packet. Only the first one is tagged as reliable (I thought anyway). ----- Original Message ----- From: "Steven Sokol" <ss...@so...> To: <iax...@li...> Sent: Thursday, February 05, 2004 9:43 AM Subject: [Iaxclient-devel] Some data related to the new bug... > 1. I cannot make the bug reproduce between my SIP client and my IAX Phone > instance. It simply does not happen. It also does not happen when calls > are placed between IAX Phone and various asterisk service (i.e. Voicemail, > MeetMe, MOH, etc.) > > 2. The audio seems to die at the point when Asterisk transmits a retry > packet: > > Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > sip*CLI> > Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 > Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] > > The CLI> above indicates where I hit the enter key just as soon as I heard > the audio die off. > > It seems the issue is specific to IAX2 and possibly to the iaxClient > library. > > Does this ring any bells for anyone? > > Thanks, > > Steve > > > > > ------------------------------------------------------- > The SF.Net email is sponsored by EclipseCon 2004 > Premiere Conference on Open Tools Development and Integration > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > http://www.eclipsecon.org/osdn > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel |
|
From: Steven S. <ss...@so...> - 2004-02-04 22:44:50
|
1. I cannot make the bug reproduce between my SIP client and my IAX Phone instance. It simply does not happen. It also does not happen when calls are placed between IAX Phone and various asterisk service (i.e. Voicemail, MeetMe, MOH, etc.) 2. The audio seems to die at the point when Asterisk transmits a retry packet: Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] sip*CLI> Tx-Frame Retry[001] -- OSeqno: 003 ISeqno: 004 Type: VOICE Subclass: 2 Timestamp: 65855ms SCall: 00006 DCall: 29893 [64.151.42.28:4569] The CLI> above indicates where I hit the enter key just as soon as I heard the audio die off. It seems the issue is specific to IAX2 and possibly to the iaxClient library. Does this ring any bells for anyone? Thanks, Steve |
|
From: Steven S. <ss...@so...> - 2004-02-04 15:13:07
|
Just wondering if the silence since my last message is due to my mail server or if everyone has left the building. Thanks, Steve |
|
From: Steven S. <ss...@so...> - 2004-02-03 02:12:10
|
Ok, looks like we have a new bug. I have experienced the same issue that several other, most recently Rattan, have mentioned; the drop off of audio after one minute. The one minute (between 1 and 2) seems to make me wonder -- could we have some problem with the registration interrupting the audio flow or otherwise trashing the stream? I say this because we re-register every minute. Could somebody with a proper Ethereal setup who is experiencing this try something: monitor the packet flow with the sniffer and see if the audio drops off after a registration (REGAUTH/REGACK) exchange. I will try to look into it from my end. Thanks! Steve |
|
From: Rattana B. <rb...@eq...> - 2004-01-31 18:32:48
|
hi, I have a little problem with IAX developpement. I use wiax.dll in my = program. When i start a comminucation with two IAX clients the communication stop = in 1'15. I mean that At this time nobody can hear voice. I use the function IAX_Call and IAX_AnswerCall ? Someone Have a tips or suggestions ? Regards Rattana |
|
From: Rattana B. <rb...@eq...> - 2004-01-31 01:00:56
|
Hi, I use wiax.dll for IAX client. I can't get the callerID from someone = call me. With a phone a can but not by my softphone. There is a way to get it ?=20 Thanks in advance Rattana |
|
From: Jarek H. <pow...@gd...> - 2004-01-28 17:15:49
|
Hi, I am new here.
I am trying to compile a little modified IaxComm (for Ansterisk in my
school) under Windows XP (i tried win2k too) with Cygwin (uname:
CYGWIN_NT-5.1 falken 1.5.5(0.94/3/2) 2003-09-20 16:31 i686 unknown unknown
Cygwin) with gcc (3.3.1) and make (3.8). I have also installed MinGW (i
tried 2.0.0-3 and 3.1.0-1) and WxWindows (i tried 2.4.1 and 2.4.2). I have
correct enviroment variables (list below). I set the "MINGW32=0" in
makeg95.env (wxwin) and "findstring CYGWIN_NT-5.1" in makefile (iaxcomm).
/sample/minimal is compiling corectly (there is some warnings). When I try
to compile IaxComm or WX (/simpleclient/wx) with make under Cygwin, i get a
lot of compile errors. I have no idea whats going on :(
There is a piece of my 2> strings.
For wxwindows (make -f makefile.g95) :
../../src/msw/listctrl.cpp:2442: warning: `wxListItemAttr*
wxGetInternalDataAttr(wxListCtrl*, long int)' defined but not used
cc1: warning: "-fno-rtti" is valid for C++ but not for C/ObjC
cc1: warning: "-fno-rtti" is valid for C++ but not for C/ObjC
cc1: warning: "-fno-rtti" is valid for C++ but not for C/ObjC
cc1: warning: "-fno-rtti" is valid for C++ but not for C/ObjC
cc1: warning: "-fno-rtti" is valid for C++ but not for C/ObjC
...
And for IaxComm (or WX, whatever):
In file included from c:/wxwindows-2.4.1/include/wx/platform.h:88,
from c:/wxwindows-2.4.1/include/wx/defs.h:23,
from c:/wxwindows-2.4.1/include/wx/wxprec.h:13,
from wx.cc:2:
c:/wxwindows-2.4.1/include/wx/chkconf.h:734:17: #error
"wxUSE_PROTOCOL_FTP/HTTP requires wxUSE_SOCKETS"
c:/wxwindows-2.4.1/include/wx/chkconf.h:1042:13: #error "wxCalendarCtrl
requires wxSpinButton and wxComboBox"
In file included from c:/wxwindows-2.4.1/include/wx/wxprec.h:13,
from wx.cc:2:
c:/wxwindows-2.4.1/include/wx/defs.h:550:10: #error "Please define a 32 bit
type"
In file included from c:/wxwindows-2.4.1/include/wx/wx.h:23,
from wx.cc:5:
c:/wxwindows-2.4.1/include/wx/event.h:837: error: syntax error before `)'
token
c:/wxwindows-2.4.1/include/wx/event.h: In member function `bool
wxKeyEvent::ControlDown() const':
c:/wxwindows-2.4.1/include/wx/event.h:823: error: `m_controlDown' undeclared
(first use this function)
c:/wxwindows-2.4.1/include/wx/event.h:823: error: (Each undeclared
identifier
is reported only once for each function it appears in.)
c:/wxwindows-2.4.1/include/wx/event.h: In member function `bool
wxKeyEvent::MetaDown() const':
c:/wxwindows-2.4.1/include/wx/event.h:824: error: `m_metaDown' undeclared
(first use this function)
...etc.
and return of set command under cygwin:
!C:='C:\cygwin\bin'
ALLUSERSPROFILE='C:\Documents and Settings\All Users'
APPDATA='C:\Documents and Settings\Powolniak\Application Data'
BASH=/usr/bin/bash
BASH_VERSINFO=([0]="2" [1]="05b" [2]="0" [3]="1" [4]="release"
[5]="i686-pc-cygw
in")
BASH_VERSION='2.05b.0(1)-release'
CLIENTNAME=Console
COLUMNS=80
COMMONPROGRAMFILES='C:\Program Files\Common Files'
COMPUTERNAME=FALKEN
COMSPEC='C:\WINDOWS\system32\cmd.exe'
CVS_RSH=/bin/ssh
DIRSTACK=()
EUID=1003
GROUPS=()
HISTFILE=/home/Powolniak/.bash_history
HISTFILESIZE=500
HISTSIZE=500
HOME=/home/Powolniak
HOMEDRIVE=C:
HOMEPATH='\Documents and Settings\Powolniak'
HOSTNAME=falken
HOSTTYPE=i686
IFS=$' \t\n'
INFOPATH=/usr/local/info:/usr/info:/usr/share/info:/usr/autotool/devel/info:
/usr
/autotool/stable/info:
LINES=25
LOGONSERVER='\\FALKEN'
MACHTYPE=i686-pc-cygwin
MAILCHECK=60
MAKE_MODE=unix
MANPATH=/usr/local/man:/usr/man:/usr/share/man:/usr/autotool/devel/man:
NUMBER_OF_PROCESSORS=1
OLDPWD=/cygdrive/c/wxwindows-2.4.1/samples/iaxclient/simpleclient
OPTERR=1
OPTIND=1
OS=Windows_NT
OSTYPE=cygwin
PATH=/usr/local/bin:/usr/bin:/bin:/usr/X11R6/bin:/cygdrive/c/WINDOWS/system3
2:/c
ygdrive/c/WINDOWS:/cygdrive/c/WINDOWS/System32/Wbem:/cygdrive/c/mingw/bin:/u
sr/b
in
PATHEXT='.COM;.EXE;.BAT;.CMD;.VBS;.VBE;.JS;.JSE;.WSF;.WSH'
PIPESTATUS=([0]="0")
PPID=1
PRINTER=
PROCESSOR_ARCHITECTURE=x86
PROCESSOR_IDENTIFIER='x86 Family 6 Model 8 Stepping 3, GenuineIntel'
PROCESSOR_LEVEL=6
PROCESSOR_REVISION=0803
PROGRAMFILES='C:\Program Files'
PROMPT='$P$G'
PS1=$'\\[\\033]0;\\w\\007\n\\033[32m\\]\\u@\\h \\[\\033[33m\\w\\033[0m\\]\n$
'
PS2='> '
PS4='+ '
PWD=/cygdrive/c/wxwindows-2.4.1/samples/iaxclient/simpleclient/wx
SESSIONNAME=Console
SHELL=/bin/bash
SHELLOPTS=braceexpand:emacs:hashall:histexpand:history:interactive-comments:
moni
tor
SHLVL=1
SYSTEMDRIVE=C:
SYSTEMROOT='C:\WINDOWS'
TEMP=/cygdrive/c/DOCUME~1/POWOLN~1/LOCALS~1/Temp
TERM=cygwin
TMP=/cygdrive/c/DOCUME~1/POWOLN~1/LOCALS~1/Temp
UID=1003
USER=Powolniak
USERDOMAIN=FALKEN
USERNAME=Powolniak
USERPROFILE='C:\Documents and Settings\Powolniak'
WINDIR='C:\WINDOWS'
WXWIN=c:/wxwindows-2.4.1
_=--version
I think that somenthing wrong with my wxwindows, but I don't have idea what.
Can You help me, please ?
Jarek
|
|
From: Adam H. <ad...@te...> - 2004-01-28 12:22:41
|
From: "Dan" <dt...@fx...> > Hi, > > ----- Original Message ----- > From: Adam Hart > > > I believe you can connect using a standard asterisk box but you'll miss > out on the extended features. > > It is a standard IAX client or a softphone using a modified/non-standard > version of IAX2? > I have tried to register with my own * box without success. It should work, others have reported success. In networks, ensure you select IAX. Firefly protocol runs over FRAME_TEXT in the iax2 protocol but only if you select Firefly as the type of network. Just a quick note, whenever I say IAX i mean iax2, iax1 isn't used at all. > > > The network runs on iLBC so unforunately it won't work with most IAX2 > clients (unless you get * to translate) > There is any specific/technical reason not to use GSM as another possible > codec? This was mainly a quality decision, only allowing iLBC ensured compability and quality. Beats asking 'which codec are you using?', 'do you have iLBC enabled?'. Also native transfers don't work between different codecs. > > > If you don't wish to connect to the firefly network, click cancel when it > asks you. > When you have setup multiple servers and try to dial a number which of the > servers is used to place that call? > I have defined just one (my box) and then tried to call an existing > extension, but it doesn't work. dialing a number without an @ suffix will use your internal network. eg 12345678 (To make a network your internal network, tick route internal calls for that network - ditto for external) dialing a number with a + at the start (and no @ suffix) will use your external network. eg +413123135 you can dial a number with @<network name> to explicitly use a certain network, eg if you have a network named iaxtel and you dial 1234@iaxtel, it will use your iaxtel network details to make the call. Otherwise, urls will work eg iax://123@my*server.com/123 - next release will add more support for urls. - push F2 to enter a url Additional - new release out Music on Hold is now working, play mp3s to other party while they are on hold :) you can now store urls in your contacts (used to limit to 20 chars, ta JTodd for finding that) |
|
From: Dan <dt...@fx...> - 2004-01-28 09:59:10
|
Hi, ----- Original Message ----- From: Adam Hart > I believe you can connect using a standard asterisk box but you'll miss out on the extended features. It is a standard IAX client or a softphone using a modified/non-standard version of IAX2? I have tried to register with my own * box without success. > The network runs on iLBC so unforunately it won't work with most IAX2 clients (unless you get * to translate) There is any specific/technical reason not to use GSM as another possible codec? > If you don't wish to connect to the firefly network, click cancel when it asks you. When you have setup multiple servers and try to dial a number which of the servers is used to place that call? I have defined just one (my box) and then tried to call an existing extension, but it doesn't work. Best regards, Dan |
|
From: Adam H. <ad...@te...> - 2004-01-28 01:14:33
|
After many months of development, I'm pleased to announced Firefly - an = IAX soft phone and network. The firefly softphone - free, runs under windows, allows connection to = multiple networks, skinable interface, connection to firefly network, = IAX2 protocol, (SIP in next release), codecs supported - iLBC, G.711 = ulaw/alaw, GSM. - contact lists, selectable ringtones. download from here - http://www.virbiage.com/firefly/ The firefly network - also free, runs on an enhanced version of IAX2 = (simply uses IAX2 text messages for customised part), voicemail, text = messaging, online presence, ability to indicate status (available, away, = NA). I believe you can connect using a standard asterisk box but you'll = miss out on the extended features. The network runs on iLBC so = unforunately it won't work with most IAX2 clients (unless you get * to = translate) Thousands of people have used it but it's still regarded in beta, as we = are still in heavy development (so expect a few bugs). It doesn't use = iaxcomm as we needed our own framework to support sip, including our own = jitterbuffer. If you don't wish to connect to the firefly network, click = cancel when it asks you. Coming soon features SIP - in alpha, few bugs outstanding music onhold - playing mp3s while the other party is onhold fast audio - will reduce the latency by 40-50ms speex - (if anyone wants it?) Feel free to contact me on or off the list to report bugs and = suggestions. I'll post everytime we release a new version (probably = every week), including fixed bugs and new features Our website is http://www.virbiage.com/ |
|
From: Mark S. <mar...@di...> - 2004-01-26 15:34:37
|
* freezes is a particularly bad thing. We need to figure that out before we fix support in iax2. Fixing it is actually not especially difficult since "POKE" Is just like "PING" only the call is terminated at the end. Mark On Mon, 26 Jan 2004, Matteo Brancaleoni wrote: > Hi. > > I was wondering if qualify support is in iaxcomm > or libiax2 ... that because if I enable qualify=yes > into iax.conf friend definition, I always see > the peer in status "UNKNOWN" and after a while > * freezes (perhaps because is sending out pings > never ack'ed ?) > > Greets, > Matteo. > > -- > Matteo Brancaleoni > Espia System Administrator > Email : mbr...@es... > Web : http://www.espia.it > Phone : +39 02 70633354 - ext 201 > IAX(2): guest@213.140.14.155 - ext 201 > Iaxtel: 1-700-56-62458 - ext 201 > > > > ------------------------------------------------------- > The SF.Net email is sponsored by EclipseCon 2004 > Premiere Conference on Open Tools Development and Integration > See the breadth of Eclipse activity. February 3-5 in Anaheim, CA. > http://www.eclipsecon.org/osdn > _______________________________________________ > Iaxclient-devel mailing list > Iax...@li... > https://lists.sourceforge.net/lists/listinfo/iaxclient-devel > |
|
From: Matteo B. <mbr...@es...> - 2004-01-26 10:41:26
|
Hi. I was wondering if qualify support is in iaxcomm or libiax2 ... that because if I enable qualify=yes into iax.conf friend definition, I always see the peer in status "UNKNOWN" and after a while * freezes (perhaps because is sending out pings never ack'ed ?) Greets, Matteo. -- Matteo Brancaleoni Espia System Administrator Email : mbr...@es... Web : http://www.espia.it Phone : +39 02 70633354 - ext 201 IAX(2): guest@213.140.14.155 - ext 201 Iaxtel: 1-700-56-62458 - ext 201 |
|
From: Michael V. D. <mv...@va...> - 2004-01-25 02:48:23
|
On Sat, 24 Jan 2004 19:37:27 -0600, Steven Sokol =
<ss...@so...>
wrote:
>Has anybody experimented with directing audio from a file (i.e. a wave =
file)
>over the IAX connection from the client end? How about directing =
incoming
>audio _to_ a file? I think Steve K had some file code but I couldn't =
figure
>out how it worked. I would like to be able to implement the following =
three
>features:
>
>1. Record both the input stream and output stream to a file (for call
>recording purposes). This would be similar to the monitor function
>available from the Manager interface (and/or CLI). I would guess that =
at
>some point in the input and output process it would be feasible to =
direct a
>copy of the data to a file. It would be great to be able to put a =
"record
>locally" feature into the phone allowing users to click to record/stop
>recording.
This sounds a feature that ought to eventually be in the library, but I =
don't
(personally) miss it yet.
>2. I would like people to play file data over the line. This could be =
used
>to play custom answering messages, and it could also be used to create a
>very cool off-board IVR functionality. Think about custom, client-side
>scripting of play/record over IAX?
I would really like to see this.
>3. Can IAX endpoints receive DTMF (or the equivalent thereof) digits?
I'd like to be able to do this. DTMF is currently rported by the library=
as
Unkown event of type 5.
A quick and dirty fix would be to send the DTMF as an iaxc_ev_text, but I=
think
that it would be better do make a new struct like:
--8<---
struct iaxc_ev_dtmf {
int callNo;
int char digit;
}
--8<----
and add it to the ev union in the iaxc_event_struct struct.
IAXC_EVENT_DTMF would be 4.
>I think you can see where I am going. It would be an extremely powerful
>extension to the library to be able to create automated telephony
>applications off-board of the Asterisk. It also seems like cool
>functionality to build into advanced soft-phones.
>
>I realize that most of you want to get the current library stable before=
we
>go hacking further functionality into it. I respect that. If you think=
I
>should hold off (or at least not tap the group's collective power) =
please
>let me know.
I'm not sure what I think on this. On one hand, I would love to see =
additional
functionality, and think that I could be of help. On the other hand, I
recognize the need to maximize the stability.
On the gripping hand, I'm not real sure that we can do much more in the =
area of
stability until the iax2 protocol white paper is published. It appears =
to me
that any problems with libiaxclient are to to a lack of understanding of =
the
iax2 protocol.
I would therefore propose that hold off on a feature freeze until the =
white
paper is published. Then we shift our attention to white paper =
compliance
Steve K, how bout I put Steve S' blind transfer stuff in CVS tomorrow, =
and we
trade diffs amongst ourselves next week while you're gone?
--
Michael (aka Steve V)
|
|
From: Steven S. <ss...@so...> - 2004-01-25 01:38:26
|
Has anybody experimented with directing audio from a file (i.e. a wave file) over the IAX connection from the client end? How about directing incoming audio _to_ a file? I think Steve K had some file code but I couldn't figure out how it worked. I would like to be able to implement the following three features: 1. Record both the input stream and output stream to a file (for call recording purposes). This would be similar to the monitor function available from the Manager interface (and/or CLI). I would guess that at some point in the input and output process it would be feasible to direct a copy of the data to a file. It would be great to be able to put a "record locally" feature into the phone allowing users to click to record/stop recording. 2. I would like people to play file data over the line. This could be used to play custom answering messages, and it could also be used to create a very cool off-board IVR functionality. Think about custom, client-side scripting of play/record over IAX? 3. Can IAX endpoints receive DTMF (or the equivalent thereof) digits? I think you can see where I am going. It would be an extremely powerful extension to the library to be able to create automated telephony applications off-board of the Asterisk. It also seems like cool functionality to build into advanced soft-phones. I realize that most of you want to get the current library stable before we go hacking further functionality into it. I respect that. If you think I should hold off (or at least not tap the group's collective power) please let me know. Thanks, Steve S. |
|
From: Steve K. <st...@st...> - 2004-01-23 23:19:40
|
Michael Van Donselaar wrote:
>On Fri, 23 Jan 2004 16:26:15 -0600, Steven Sokol <ss...@so...>
>wrote:
>
>
>
>>I have also added a nifty little hold/unhold function that adds two
>>functions to iaxClient and one to libiax2. The libiax2 function allows for
>>quelch/unquelch with an optional MOH IE appended to the command request.
>>This allows you to place a call on hold and to "recall" the call without
>>having to switch to a different appearance (call). It also gives you a way
>>to provide MOH to the other active-but-held calls when you _do_ switch to
>>another appearance.
>>
>>It's working fine in early testing now. It may appear as a feature in IAX
>>Phone this weekend.
>>
>>Do you guys want it as a standard part of the lib? Is there a way to create
>>an IE and send it in a command message w/o requiring changes to
>>libiax2/iax.c?
>>
>>
>
>You can already do that by
>
>iaxc_select_call(-1);
>
>I have a local change to iaxclient_lib.c that sends a message unselecting the
>selected call. It's missing from the CVS source. I've been meaning to update
>it for two days.
>
>My solution was a kludge. A new function may be more elegant.
>
>
Steve V,
I think that Steve S' solution might be better or more correct,
because it tells the other side the call is on hold, and they get music
and whatnot. (I don't use this stuff, so I don't know how it works,
really).
What I'd suggest for implementing this, though, is to use Steve V's
API (selecting -1), but change the library such that all calls on hold
(active but unselected) get the quelch/MOH treatment.
As to Steve S' second question (about sending an IE along without
modifying libiax2), I don't know, but I'd prefer not to have to modify
libiax2 more than necessary.
Also, as a note: I'll be away next week, but I'll read e-mail occasionally.
-SteveK
|
|
From: Michael V. D. <mv...@va...> - 2004-01-23 22:58:42
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On Fri, 23 Jan 2004 16:26:15 -0600, Steven Sokol = <ss...@so...> wrote: >I have also added a nifty little hold/unhold function that adds two >functions to iaxClient and one to libiax2. The libiax2 function allows = for >quelch/unquelch with an optional MOH IE appended to the command request. >This allows you to place a call on hold and to "recall" the call without >having to switch to a different appearance (call). It also gives you a = way >to provide MOH to the other active-but-held calls when you _do_ switch = to >another appearance. > >It's working fine in early testing now. It may appear as a feature in = IAX >Phone this weekend. > >Do you guys want it as a standard part of the lib? Is there a way to = create >an IE and send it in a command message w/o requiring changes to >libiax2/iax.c? You can already do that by=20 iaxc_select_call(-1); I have a local change to iaxclient_lib.c that sends a message unselecting= the selected call. It's missing from the CVS source. I've been meaning to = update it for two days. My solution was a kludge. A new function may be more elegant. -- Steve V :) |
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