October 2011 - freeswitch-dev@lists.freeswitch.org

Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 64, Issue 19
by marco blaisou 24 Oct '11

24 Oct '11

COMPILING FREESWITCH ON WINDOWS 7 FROM THE GIT ----------------------------------------------------------------------------------------- Hi guys, just to give you some update from the trenches and give you some update on compiling FreeSwitch on Windows from the git.(what a journey!!!) I can tell you that so far, it appeared to be one of the most frustrating experience in building a system from a git: files are missing or are not loaded properly as they should... So after a few nights of sctatching my head, here is the status: *Building on Windows from a git repository -------------------------------------------------------------* After git cloning the repository, open vs2008 solution then: Stop Comodo or any firewall and virus detection (if not this screws up download scripts giving: util.vbs(184, 2) msxml3.dll errors) *0) launch build* *1) insure all the supposed dowloaded directories are present. (!!!!!!!!!!!!! VITAL !!!!!!!!)* Actually there are some problems: -> issue with opensssl download ------------------------------------------------- for a strange reason after autodownload the untar is not complete, resulting in missing subdirectory So make sure to do it by hand if necessary -> issue with CELT download ------------------------------------------- MUST change prebuilt instructions to : if not exist "$(ProjectDir)..\celt-0.10.0" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/celt-0.10.0.tar.gz"$(ProjectDir).." xcopy "$(ProjectDir)\celt\config.h" "$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I xcopy "$(ProjectDir)\celt\float_cast.h" "$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I *2) if you uses Windows sdk 7.1* edit fake-rfc2533.h and comment out : #ifdef nodef // *****add this #ifndef HAVE_STRUCT_ADDRINFO struct addrinfo { int ai_flags; /* AI_PASSIVE, AI_CANONNAME */ int ai_family; /* PF_xxx */ int ai_socktype; /* SOCK_xxx */ int ai_protocol; /* 0 or IPPROTO_xxx for IPv4 and IPv6 */ size_t ai_addrlen; /* length of ai_addr */ char *ai_canonname; /* canonical name for hostname */ struct sockaddr *ai_addr; /* binary address */ struct addrinfo *ai_next; /* next structure in linked list */ }; #endif /* !HAVE_STRUCT_ADDRINFO */ #endif // *****add this *3) if issue with folder: sofia-sip * cannot include file sip-extra.h ??????????????????????????????????????? (I'm fed up with this so I just decided to get it from a tar) *4) pb on compile : libsndfile* in sofia_presence.c if (mod_sofia_globals.debug_presence > 1) { int i; // to be added for (i = 0; i < argc; i++) { //---------------------------------------------------------------------------------- That's it for the time being, most of the stuff (but not ALL) now compile and I'll try to focus on other pending issues. OBVIOUSLY ANY ADVICE IS MORE THEN WELCOME !!!! 2011/10/23 <freeswitch-dev-request(a)lists.freeswitch.org> > Send FreeSWITCH-dev mailing list submissions to > freeswitch-dev(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > or, via email, send a message with subject or body 'help' to > freeswitch-dev-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-dev-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-dev digest..." > > Today's Topics: > > 1. == ISSUE building FreeSwitch on Windows (marco blaisou) > 2. Re: Strange behaviour in SOFIA SIP (Visciano Giovanni) > 3. Re: == ISSUE building FreeSwitch on Windows (Peter Olsson) > > > ---------- Message transféré ---------- > From: marco blaisou <blaisounou(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org > Date: Sat, 22 Oct 2011 16:59:56 +0200 > Subject: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows > Hi guys, > For the few days I've been unsuccesfull in building FreeSwith on Windows 7 > > I did follow the Wiki line per line, but some thing is really going wrong: > > apparently the files in /libs directory are not loaded from the net (ie > the util.vbs does seem to load and unpack the libs) > error is : *freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le système > ne trouve pas la ressource spécifiée (ie system doesn't find specified > resource)* > > > then obviously the compiler starts complaining about missing stuff such > as: > *c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) > : fatal error C1083: Cannot open include file: 'switch_version.h': No such > file or directory* > > To summarize: > I loaded the FreeSwitch git repository with TortoiseGit > > My config is : > Windows 7/64bits > VSC2008 sp1 > Windows SDK 7.1 > > My firewall is desactivated as suggested > > I did follow the recommendation to clear some files: > *del libs\apr\include\apr.h > del libs\js\config.h > del libs\js\src\jsautocfg.h > del libs\js\nsprpub\pr\include\prcpucfg.h > del libs\iksemel\include\config.h > del libs\xmlrpc-c\xmlrpc_config.h > del libs\libsndfile\src\sfconfig.h > del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h > del libs\js\nsprpub\pr\include\prcpucfg.h > > cd libs\win32\libsndfile > rename config.h sfconfig.h* > > Could a guru provide some hints on how to solve this??? > Thanks > > > > > ---------- Message transféré ---------- > From: "Visciano Giovanni" <Giovanni.Visciano(a)italtel.it> > To: <freeswitch-dev(a)lists.freeswitch.org> > Date: Sat, 22 Oct 2011 20:30:06 +0200 > Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP > Now I understand. > If both legs use the same codec, the codec ("fake") will be used in a > pass-through way, > so no additional transcoding is applied, as in PROXY PASS-THROUGH > condition. > > So let me reason in a default media scenario (no proxy-media). > > Now, as I explained in the first mail, sofia build "ep_codec_string" > (SWITCH_ > ORIGINATOR_CODEC_VARIABLE) as the intersection between the codec list > received > in SDP offer and the codec loaded in the core. > So if the offered SDP codec list is NOT a subset of the loaded core codecs > the other > legs will miss some codecs proposed by the calling endpoint. > That said, if I want to be sure every codecs my endpoint support won't be > filtered I > have to register a "fake" pass-through codec for each one. > > ... you did not answer to my last question :). > I haven't found any other endpoint module the can interwork in PROXY mode > with > sofia (even h323 doesn't support this). My question: is there a standard/ > recommended way to exchange capability between endpoint of different nature > in FS > proxy mode? Maybe using SDP o/a as done in sofia sip? > > Thank you again for you help. > Giovanni > > > -----Original Message----- > From: freeswitch-dev-bounces(a)lists.freeswitch.org on behalf of Anthony > Minessale > Sent: Sat 10/22/2011 4:48 PM > To: freeswitch-dev(a)lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP > > that's what I was trying to explain. > PROXY mode does not let you manipulate anything, it's pure > pass-through negotiation between the endpoints. > > All the stuff with ep_codec_string are relevant only when you are not > trying to use proxy mode. > This is a common mistake people make assuming not using proxy mode > means somehow doing more work but it's not the case, any time both > legs of a call use the same codec it's equally efficient to what you > end up with in proxy mode. proxy mode simply means take the inbound > SDP and offer it AS IS to the far end and vice-versa and use a fake > codec in the middle that is agnostic. > > > > On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni > <Giovanni.Visciano(a)italtel.it> wrote: > > Hello Anthony. > > Maybe I want not so clear, let me explain. > > > > I do not want to apply any codec manipulation, I'm working on a prototype > endpoint module that has "proxy media capabilty" and wants to interoperate > with other endpoints. > > What I saw is that in a sofia vs sofia call (proxy mode+late negotiation) > they exchange through FS calling/called endpoint capability looking at SDP > session carried by channel variables, so internally they use SDP O/A scheme. > > I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could > be used to exchange capability without using a protocol specific format > (SDP). > > If this is not the case, what's the right way to negotiate capability > between endpoints in proxy mode? > > > > Than you. > > Giovanni > > > > PS: I did not find any existing endpoint except sofia that support this. > Am I right? > > > > -----Original Message----- > > From: freeswitch-dev-bounces(a)lists.freeswitch.org on behalf of Anthony > Minessale > > Sent: Fri 10/21/2011 5:06 PM > > To: freeswitch-dev(a)lists.freeswitch.org > > Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP > > > > any codec manipulation techniques and proxy media mode are not > > compatible. Proxy media mode means stay completely out of the codec > > negotiation process. > > > > On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni > > <Giovanni.Visciano(a)italtel.it> wrote: > >> Preconditions on sip profile > >> ---------------------------- > >> inbound-codec-prefs NOT set > >> outbound-codec-prefs NOT set > >> inbound-proxy-media=TRUE > >> inbound-bypass-media=FALSE > >> inbound-late-negotiation=TRUE + inherit_codec=TRUE > >> > >> TEST scenario > >> ------------- > >> SIP phone send INVITE with SDP to FS > >> > >> INVITE > >> SDP[codelist=PCMA, PCMU, G722, G729] --> FS > >> > >> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, > L16] > >> > >> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU] > >> > >> EXPECTED > >> -------- > >> In "proxy-media + late-negotiation" scenario: > >> ep_codec_string=[PCMA,PCMU,G722,G729] > >> > >> PROBLEM > >> ------- > >> According to the documentation > >> (http://wiki.freeswitch.org/wiki/Codec_negotiation) > >> > >> "ep_codec_string: This variable is only available if late negotiation is > >> enabled on the profile. It's a readable string containing all the codecs > >> proposed by the calling endpoint." > >> and > >> "inherit_codec=true (only applicable when late-negotiation is enabled) > will > >> take the codec negotiated when the B leg answers and pass it to the A > leg so > >> it also uses the same codec (if the A leg supports it); otherwise, it > will > >> use whatever it can from its own list" > >> > >> The variable ep_codec_string doesn't contain ALL the codecs proposed by > the > >> calling endpoint. It contains the intersection between the codecs > proposed > >> by the calling and the codecs loaded in FS. > >> In proxy-media / no-media configuration the called endpoint doesn't have > the > >> calling codec list so he can't negotiate using all the capabilities > >> available. > >> (I'm working on a test endpoint module that should work in proxy and > >> no-media configuration ... mod_sofia vs mod_sofia works because they > look at > >> the remote sdp variable) > >> > >> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string() > >> > >> > >> Regards, > >> Giovanni > >> > >> > >> > >> Internet Email Confidentiality Footer > >> > >> > ******************************************************************************************************************************************** > >> > >> La presente comunicazione, con le informazioni in essa contenute e ogni > >> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' > >> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non > siete > >> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, > >> comunicazione, divulgazione o simili basate sul contenuto di tali > >> informazioni e' vietata e potrebbe essere contro la legge (art. 616 > C.P., > >> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). > Se > >> avete ricevuto questa comunicazione per errore, vi preghiamo di darne > >> immediata notizia al mittente e di distruggere il messaggio originale e > ogni > >> file allegato senza farne copia alcuna o riprodurne in alcun modo il > >> contenuto. > >> > >> This e-mail and its attachments are intended for the addressee(s) only > and > >> are confidential and/or may contain legally privileged information. If > you > >> have received this message by mistake or are not one of the addressees > >> above, you may take no action based on it, and you may not copy or show > it > >> to anyone; please reply to this e-mail and point out the error which has > >> occurred. > >> > ******************************************************************************************************************************************** > >> > >> _______________________________________________ > >> Join us at ClueCon 2011, Aug 9-11, Chicago > >> http://www.cluecon.com 877-7-4ACLUE > >> > >> FreeSWITCH-dev mailing list > >> FreeSWITCH-dev(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > >> http://www.freeswitch.org > >> > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org > > googletalk:conf+888@conference.freeswitch.org > > pstn:+19193869900 > > > > _______________________________________________ > > Join us at ClueCon 2011, Aug 9-11, Chicago > > http://www.cluecon.com 877-7-4ACLUE > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > _______________________________________________ > > Join us at ClueCon 2011, Aug 9-11, Chicago > > http://www.cluecon.com 877-7-4ACLUE > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > ---------- Message transféré ---------- > From: Peter Olsson <peter.olsson(a)visionutveckling.se> > To: "freeswitch-dev(a)lists.freeswitch.org" < > freeswitch-dev(a)lists.freeswitch.org> > Date: Sun, 23 Oct 2011 08:42:42 +0200 > Subject: Re: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows > I believe that you might be missing some system dll's on your machine. > > msxml3.dll is something that should exist on your computer, and I think the > download script is depending on it. > > /Peter > ________________________________________ > Från: freeswitch-dev-bounces(a)lists.freeswitch.org [ > freeswitch-dev-bounces(a)lists.freeswitch.org] för marco blaisou [ > blaisounou(a)gmail.com] > Skickat: den 22 oktober 2011 16:59 > Till: freeswitch-dev(a)lists.freeswitch.org > Ämne: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows > > Hi guys, > For the few days I've been unsuccesfull in building FreeSwith on Windows 7 > > I did follow the Wiki line per line, but some thing is really going wrong: > > apparently the files in /libs directory are not loaded from the net (ie > the util.vbs does seem to load and unpack the libs) > error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le système ne > trouve pas la ressource spécifiée (ie system doesn't find specified > resource) > > > then obviously the compiler starts complaining about missing stuff such > as: > c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) > : fatal error C1083: Cannot open include file: 'switch_version.h': No such > file or directory > > To summarize: > I loaded the FreeSwitch git repository with TortoiseGit > > My config is : > Windows 7/64bits > VSC2008 sp1 > Windows SDK 7.1 > > My firewall is desactivated as suggested > > I did follow the recommendation to clear some files: > del libs\apr\include\apr.h > del libs\js\config.h > del libs\js\src\jsautocfg.h > del libs\js\nsprpub\pr\include\prcpucfg.h > del libs\iksemel\include\config.h > del libs\xmlrpc-c\xmlrpc_config.h > del libs\libsndfile\src\sfconfig.h > del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h > del libs\js\nsprpub\pr\include\prcpucfg.h > > cd libs\win32\libsndfile > rename config.h sfconfig.h > > Could a guru provide some hints on how to solve this??? > Thanks > > > !DSPAM:4ea3271432761137115228! > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > >

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[Freeswitch-dev] trouble on how to remove the media bug from the session after the sessions are closed
by Han Xin 24 Oct '11

24 Oct '11

Hello everyone! I am new to Freeswitch and try to add media bugs such as digital signal processing to the read stream and write stream. But I have some trouble on how to remove the media bug from the session after the sessions are closed. I tried to look into other existed modules such as mod_soundtouch and mod_dptools to find a solution, the former mod_soundtouch put the function "switch_core_media_bug_remove()" in the beginning of application run function. The latter mod use a single app called "remove bugs" and put this app in the dial plan to remove the bugs. The question is where should I put this "switch_core_media_bug_remove()" function after the sessions are closed. Is there a way to automatically remove a media bug when a call is ended? Or it is only possible to use a single app to remove the media bugs as in the mod_dptools? Thank you all in advance! --

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[Freeswitch-dev] [patch] Better Python Debug Output
by STIEG Heimo 24 Oct '11

24 Oct '11

Hello, I've modified the mod_python module to create meaninfull Error Messages: Instead of this: 2011-10-19 08:28:29.356605 [ERR] mod_python.c:224 Error importing module We have now this: 2011-10-19 08:28:29.356605 [ERR] mod_python.c:224 Error importing module 2011-10-19 08:28:29.356605 [ERR] mod_python.c:156 Python Error in script nxo_transferifgroup: <type 'exceptions.AttributeError'>: 'module' object has no attribute 'argv' Exception: None Traceback: File: /usr/local/freeswitch/scripts/nxo_transferifgroup.py Line: 6 Can someone import the patch to the trunk please? http://jira.freeswitch.org/browse/FS-3615 Regards Heimo Stieg

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[Freeswitch-dev] FreeTDM delay reconnecting E1/T1
by Glaucio Vaz de Mello 24 Oct '11

24 Oct '11

Hi, I have a question for FreeTDM experts. When I have a T1/E1 ISDN-PRI link connected through Sangoma card, I can check signalling and physical status using this command: > ftdm sangoma_isdn show_spans wp1 span:wp1 physical:OK signalling:UP Unplugging E1/T1 cable some alarms are raised, but show_spans command still returns "OK" and "UP". After some time (more than one minute) I get the disconnected status: > ftdm sangoma_isdn show_spans wp1 span:wp1 physical:ALARMED signalling:DOWN Using sangoma wanpipemon command I can get disconnected status immediately: # wanpipemon -i w1g1 -c Ta Loss of Signal: ON My question is why FreeTDM takes this big delay to detect a disconnected cable? Is there a way to reduce this time? The same happens when I reconnect the cable, FreeTMD takes a minute or more to detect it. thanks, Glaucio

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[Freeswitch-dev] Wanpipe timing source - clock sharing
by Juraj Fabo 23 Oct '11

23 Oct '11

Hi I'm not sure whether this is a proper forum, but I have an issue with wanpipe/dahdi driver and the sangoma a102de card. My problem is probably the timing source change in case when the E1 cable is disconnected from the card span or spans. My application plays a PRI_CPE side and uses a testpattern/keepalive sending on one dedicated e1 channel to detect E1 breakdown. When there is no cable connected to particular span, keepalive algorithm send patterns and reads 'dummy' data so it finds out that E1 is broken. Actually I've read the info from wanpipe.spec about the free run mode in firmware version V36, I use V37, here is my configuration sles 11 SP1, wanpipe-3.5.23 (tested also 3.5.18), dahdi-2.5.0.1 (tested also 2.3.0.1), sangoma a102de, HWEC and DTMF detection off. I have an experience with digium E1 cards and DAHDI driver that removing a cable from E1 span makes the span to receive a clock from another span connected to telko. Or, given there is no other span connected to telko, to use internal clock on all spans. So on the digium with dahdi the clock source is moving from span to span as specified in "timing source" parameter of /etc/dahdi/system.conf (and indicated as MASTER in the 1st line of /proc/dahdi/X) I tested sangoma card together with wanpipe and dahdi drivers. Here the behaviour was rather different. When cable was removed, following messages were logged (amongst others) Oct 21 09:34:06 v184 kernel: [ 3185.691989] wanpipe2: Lost of Signal is detected! Oct 21 09:34:10 v184 kernel: [ 3189.604230] wanpipe2: LOF : ON Oct 21 09:34:10 v184 kernel: [ 3189.604234] wanpipe2: E1 disconnected! Oct 21 09:34:10 v184 kernel: [ 3189.604263] wanpipe2: AFT communications disabled! (Dev Cnt: 2 Cause: Link Down) Between time 09:34:06 and 09:34:10 my application still writes/reads to the channel on span which was removed. After 09:34:10 it is not possible to read/write anymore until the cable is connected back (I use ACE framework event handler registered in reactor with READ | WRITE | EXCEPT mask) When last cable is removed (tested also with quad card) then the first span 'recovers' and read/write is possible on it. Connecting one span 'enables' reading/writing on re-connected span. I think this might be a timing source issue. My question is, how to configure wanpipe/dahdi in order to share the clock on all spans of sangoma, also when there are spans which are configured and used by an application but not connected. Thank you in advance Juraj Fabo

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Re: [Freeswitch-dev] MTU setting and application buffer size
by Juraj Fabo 23 Oct '11

23 Oct '11

Hi thank you for your response. Here is my code used for testing the latency http://pastebin.com/mwHrRDeK. It is based on and built as freeswitch-git/libs/freetdm/src/testapp.c For the measurement I used a104d card with interconnected spans 1&2. App loads binary file "sample_binin" which was previously created by dumping the E1 channel when there was active call. This dump is used as a source of voice. App initiates a call on span1. App handles it as incoming call on span2. After call is established, measurement loop is entered,measurement is in single thread world, handling the spans 1 & 2 in "serial". App feeds prebuffered voice to outgoing call on span1 and in paralell makes a dump of written data to binary file "owbinout". At the same time, data read from the span1 are dumped to binary file "orbinout". Then span2 is handled, data read from the active channel are just written back, here the loopback is created. In paralell data are dumped to file "awbinout" So then results are being calculated by comparing the content of : owbinout against awbinout = one way delay owbinout against orbinout = roundtrip delay Personally I use Audacity to import the binary voice, decode it and pair in stereotracks so I can measure the delay. With this approach I got the results posted earlier. With best regards Juraj Fabo > ---------- Forwarded message ---------- > From: Moises Silva <moises.silva(a)gmail.com> > To: freeswitch-dev(a)lists.freeswitch.org > Date: Mon, 17 Oct 2011 12:22:58 -0400 > Subject: Re: [Freeswitch-dev] MTU setting and application buffer size > On Tue, Oct 4, 2011 at 8:56 AM, Juraj Fabo <juraj.fabo(a)gmail.com> wrote: > >> > Please, have a look at the results with various parameters used: >> > mtu:80 txqueue_size=1 rxqueue_size=1 one-direction delay=60ms >> > round-trip delay=120ms >> > mtu:80 txqueue_size=2 rxqueue_size=2 one-direction delay=80ms >> > round-trip delay=160ms >> > mtu:80 txqueue_size=10 rxqueue_size=10 one-direction delay=240ms >> > round-trip delay=480ms >> > mtu:40 txqueue_size=1 rxqueue_size=1 one-direction delay=40ms >> > round-trip delay=80ms >> > mtu:40 txqueue_size=2 rxqueue_size=2 one-direction delay=60ms >> > round-trip delay=120ms >> > mtu:40 txqueue_size=5 rxqueue_size=5 one-direction delay=120ms >> > round-trip delay=240ms >> > mtu:16 txqueue_size=1 rxqueue_size=1 one-direction delay=30ms >> > round-trip delay=60ms >> > mtu:8 txqueue_size=1 rxqueue_size=1 one-direction delay=25ms >> > round-trip delay=50ms >> > mtu:8 txqueue_size=10 rxqueue_size=10 one-direction delay=205ms >> > round-trip delay=410ms >> > >> >> >> Updated test result with the codec_ms=10 is following: >> mtu:16 txqueue_size=1 rxqueue_size=10 one-direction delay=20ms >> round-trip delay=50ms >> mtu:80 txqueue_size=1 rxqueue_size=10 one-direction delay=56ms >> round-trip delay=110ms >> >> However, this is still not 10ms with the default MTU:80 and codec_ms:10 :( >> >> Anyway, is it a good idea to use so assymetric tx/rx queue sizes? >> >> > In general I don't think is a good idea because 99% of the user population > is using default symmetric queues, moving to obscure/rarely-tested > configurations increases your likelihood of facing problems. Having said > that, I don't see why it should not work :-) > > Can you share the code you're using to measure latency? may be setup a git > repo in github with your code? > > *Moises Silva > **Software Engineer, Development Manager*** > > msilva(a)sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> > > Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…> > | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…> > | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…> > | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> > | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…> > | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…> > | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…>`| > | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=…> > > VegaStream is now part of Sangoma!<http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acq…> > > Ask us about both Gateway Appliances<http://sangoma.com/products/media_gateways/vega.html> > and Internal Gateways<http://sangoma.com/products/software_products/netborder_suite/netborder_exp…> > > >

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[Freeswitch-dev] unresolved external symbol _TPTAG_CAPT and _NTATAG_SRV_503 ?
by marco blaisou 23 Oct '11

23 Oct '11

as a follow up of previous mail 21> Creating library Win32\Release/mod_sofia.2008.lib and object Win32\Release/mod_sofia.2008.exp 21>mod_sofia.obj : error LNK2001: unresolved external symbol _TPTAG_CAPT 21>sofia.obj : error LNK2001: unresolved external symbol _NTATAG_SRV_503 21>C:\Users\jc\Desktop\VOIP\freeswitch\freeswitch\Release/mod/mod_sofia.dll : fatal error LNK1120: 2 unresolved externals

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[Freeswitch-dev] == ISSUE building FreeSwitch on Windows
by marco blaisou 23 Oct '11

23 Oct '11

Hi guys, For the few days I've been unsuccesfull in building FreeSwith on Windows 7 I did follow the Wiki line per line, but some thing is really going wrong: > apparently the files in /libs directory are not loaded from the net (ie the util.vbs does seem to load and unpack the libs) error is : *freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le système ne trouve pas la ressource spécifiée (ie system doesn't find specified resource)* > then obviously the compiler starts complaining about missing stuff such as: *c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) : fatal error C1083: Cannot open include file: 'switch_version.h': No such file or directory* To summarize: I loaded the FreeSwitch git repository with TortoiseGit My config is : Windows 7/64bits VSC2008 sp1 Windows SDK 7.1 My firewall is desactivated as suggested I did follow the recommendation to clear some files: *del libs\apr\include\apr.h del libs\js\config.h del libs\js\src\jsautocfg.h del libs\js\nsprpub\pr\include\prcpucfg.h del libs\iksemel\include\config.h del libs\xmlrpc-c\xmlrpc_config.h del libs\libsndfile\src\sfconfig.h del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h del libs\js\nsprpub\pr\include\prcpucfg.h cd libs\win32\libsndfile rename config.h sfconfig.h* Could a guru provide some hints on how to solve this??? Thanks

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[Freeswitch-dev] Call-ID formatting
by Kurtis Heimerl 22 Oct '11

22 Oct '11

According to the sip standard, call-ids should have a "host" trailer (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6(a)foo.bar.com) Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12 FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the above. Is there any reason why this is the case? This seems too large of an issue to be a bug or oversight. Here's an example from a FS MESSAGE: Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5

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[Freeswitch-dev] Calling Python Application from Chatplan (mod_sms help)
by Kurtis Heimerl 21 Oct '11

21 Oct '11

I'm pushing this to FreeSwitch-dev as mod_sms is still under active development. Any way to call a Python Application from the chatplan? This works: <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/> But I can't monkey with the session variables with the fs_api call. Instead, I want to do this: <action application="python" data="OpenBTS_Parse_SMS"/> and this fails in the following way: 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid chat application interface [python]! The mod_sms page seems to indicate that lua works just fine from the chatplan... is this just not implemented yet? Can I do that? Thanks!

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