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Quality headset regarding voip
by EL 22 Nov '13

22 Nov '13

Dear FS community, Can anyone advise me some quality headsets regarding voip. Real good speakers for listening music and such is also welcome, but the most important factor is quality of microphone and sound regarding voip. Anyone else who is mainly working with a softphone (or a headset combined with a normal physical phone)? There are so many manufacturers out there, who claim HD, etc., but I think most of it is just branding / marketing. Requirements or would be nice to have: - Wired (preferable with detachable / replaceable wires). - Noise Cancelling Microphone. - Doesn't rely on additional manufacturer software. - Good performance on BSD/Linux (my platform is 100% BSD/Linux). - Integrated good USB sound card, which is easier to handle on my platform(?) Price: not important, although the differences in prices are huge. Some have a focus for gaming, while others are pointing towards office environments. Example office (expensive): VXI UC ProSet LUX 31 http://www.vxicorp.com/products/contact-center-and-office-solutions/corded/… Example gaming (expensive): http://www.soundblaster.com/products/Sound-Blaster-EVO-ZxR.aspx If someone can help me/others decide what to look for: many thanks for reading and sharing your knowledge and experience with me/us! -- EL

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Re: [Freeswitch-users] Configure Freeswitch as an IMS application server
by Thomas Titty 22 Nov '13

22 Nov '13

Thanks Michael. That worked. But I ran into another issue. I am trying to make FS work as an application server in the IMS network. The IMS network while sending an INVITE to FS include a route header with Original Dialog Identifier (ODI) to indicate that the INVITE should be send back to the IMS network by FS. IMS network uses ODI to correlate the two INVITEs. When I use the dialplan with fs_path, I do not see the Route header in the terminating leg INVITE sent by FS. Is there a way to make FS function this way? Or Is there a way to copy a SIP route header from the originating leg to the terminating leg of a call? Thanks, Thomas From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, 21 November 2013 1:43 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Configure Freeswitch as an IMS application server https://wiki.freeswitch.org/wiki/Sofia-SIP#Specifying_SIP_Proxy_With_fs_path On Nov 21, 2013, at 3:58 PM, Thomas Titty <ttitty(a)kineto.com<mailto:ttitty@kineto.com>> wrote: I tried all of the variables (sip_network_destination, sip_route_uri, sip_invite_req_uri) that you suggested but the SIP INVITE is not being routed to the p-cscf. I have 2 sip users (say) user-a(a)example.com<mailto:user-a@example.com> and user-b(a)example.com<mailto:user-b@example.com>, and P-CSCF at p-cscf.com<http://p-cscf.com/> I would like the SIP INVITE from FS to look like INVITE sip:user-b@example.com SIP/2.0 From: "user-a" <sip:user-a@example.com>;tag=a7c50F74399yc To: <sip:userba@p-cscf.com:5060> Here is what my dial-plan looks like <extension name="pcscf_outbound"> <condition field="destination_number" expression="^(\d{10})$"> <action application="bridge" data="{sip_network_destination=$1(a)example.com<mailto:sip_network_destination=$1@example.com>,sip_route_uri=sip:$1@example.com<mailto:1@example.com>,sip_invite,sip_invite_req_uri=sip:$1@example.com<mailto:1@example.com>}sofia/external/$1(a)p-cscf.com<mailto:sofia/external/$1@p-cscf.com>:5060"/> </condition> </extension> With this setting the SIP INVITE looks okay, but is being sent to example.com<http://example.com/> rather than p-cscf.com<http://p-cscf.com/>. From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf OfKen Rice Sent: Thursday, 21 November 2013 10:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Configure Freeswitch as an IMS application server In the dialplan you just need to have FS send the invite back to the S-CSCF for user B... This can be done a couple of ways, 1 is specify the URI for the user@s-cscf or to specify to route it via the s-cscf there are a few ways to choose from on this... See http://wiki.freeswitch.org/wiki/Variable_sip_network_destination for 1 way with links to others On 11/20/13 11:04 PM, "Thomas Titty" <ttitty(a)kineto.com<x-msg://27/ttitty@kineto.com>> wrote: Hello, I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network. Here is the high-level signaling flow that I am considering. 1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch) 2. User-A makes a voice call to User-B. 3. S-CSCF forwards the SIP INVITE to Freeswitch. 4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF. 5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction. I went through the dial-plan documentation and could not figure out whether such a configuration is possible. Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated. Thanks, Thomas ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<x-msg://27/consulting@freeswitch.org> http://www.freeswitchsolutions.com<http://www.freeswitchsolutions.com/> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com<http://www.cudatel.com/> Official FreeSWITCH Sites http://www.freeswitch.org<http://www.freeswitch.org/> http://wiki.freeswitch.org<http://wiki.freeswitch.org/> http://www.cluecon.com<http://www.cluecon.com/> FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<x-msg://27/FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org<http://www.freeswitch.org/> -- Ken http://www.FreeSWITCH.org<http://www.freeswitch.org/> http://www.ClueCon.com<http://www.cluecon.com/> http://www.OSTAG.org<http://www.ostag.org/> G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net<http://irc.freenode.net> #freeswitch _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] Freeswitch Performance
by Andre Demattia 22 Nov '13

22 Nov '13

I had debug turned off but I'll enable it to see if anything interesting happens. I didn't see anything that said pause. I'll also turn off odbc for the core db and use the default one to see if that helps. Un testing it didn't help but I'll try it in again. -----Original Message----- From: "Steven Ayre" <steveayre(a)gmail.com> Sent: ‎11/‎20/‎2013 4:36 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Freeswitch Performance > > show nothing other than FreeSWITCH was receiving the new messages but no > bridge or cancel actions were taking place Do you have your logs at debug level and are there any log messages that indicate a call reaching a point then pausing? I saw something recently similar under load, and it turned out to be the limit subsystem was waiting on DB queries that were severely backing up because there were too many commits happening per second and InnoDB was configured to flush to disk on every transaction commit which can only happen ~200 times per second. (Switching to the memory engine fixed the error although isn't transactional). -Steve -Steve On 20 November 2013 20:38, Andre <andretodd(a)verizon.net> wrote: > I tested again and when I use 25 cps for 2 hours everything worked fine. > When I sent 50 cps I noticed FreeSWITCH worked for about 5 minutes then I > saw the memory jump from 122 mb to to 1.6 gb in seconds. I think FreeSWITCH > crashed but the logs show nothing other than FreeSWITCH was receiving the > new messages but no bridge or cancel actions were taking place. > > > > How do I tell what happened that made freeswitch just stop working? The > GUI didn’t show anything like a throw exception and when I type version or > sofia status in the FreeSWITCH gui it doesn’t respond. > > > > Andre > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, November 20, 2013 12:33 AM > *To:* Freeswitch-users > > *Subject:* Re: [Freeswitch-users] Freeswitch Performance > > > > Get HEAD of stable branch or HEAD of master for latest build files. > > There is no promise windows can do 500cps the industry standard is 50 and > you'll have probably more luck on linux but there is only on way to fimd > out. > > On Nov 19, 2013 11:27 PM, "I put the Who? in Mishehu" < > mishehu(a)freeswitch.org> wrote: > > The amusing thing about the recurring load test threads that pop up every > now and then is that the most important tips are already documented on the > wiki in one single page. Beyond that, you probably spend more time load > testing than you do just tossing up another server and using something like > kamailio or opensips to load balance... > > > -- > > Yossi Neiman > > > > > > On 11/19/2013 06:48 PM, Antonio Teixeira wrote: > > Hello Andre. > > Solving performance problems is always hard , but a few points to keep in > mind : > > To my knowledge FS is pretty much a linux software , it works on windows > by that i doesn't means it performs on windows ( please correct me if i'm > wrong) > The native DB for FS is Postgresql ( this should make you oracle admin > more happy :) ). > > Try this : > Linux Server Realtime Kernel + Last FS Stable + Postgresql ( tune it well). > Take care with ulimit on linux , its needed for the extra punch of > performance > Check if your "performance test" is real if you are pointing to 1500CPS > for a two minute call you will have 180k calls before the first call gets > disconnected, if you are also testing RTP that means 180k * 2 ports needed > , The Kernel will start to complaining and you will run out of ports you > will also run out of FD probably before you run out of ports on a not tuned > linux install. > If the DB is on the same server it will also stall since the FD will not > be present for new connections to the back end. > > In terms of BW you will exceed 230000 kb/s add a few more kb/s for the DB > ( Not a problem here i think) > > Another thing , no sip server ( please include all the nice names here > like B2BUA , etc ) should ever queue anything. > I have never read the code that handles this on FS but the way i would do > it should be something like : can we sustain the load ? No in that case > send 503 else proceed. > This is done in freeswitch by this var > https://wiki.freeswitch.org/wiki/XML_Switch_Configuration#max-sessions > > A more general approach: > In SIp Servers you will generally ( if not running cpu intensive tasks, > trancoding , tone detection , etc ), consume all the ram , Network IO ,than > the cpu. > For the guys that plan something big you should take a look at this : > http://www.solarflare.com/High-Frequency-Trading > Not by The NIC itself but because of this : > http://www.openonload.org > > Without more data i don't think we can help , above all performance > testing is a dark art you just need to keep on trying different things and > above all have some type of monitor tool to understand what exactly is > holding you up , maybe its no FS... > > Also 1500CPS is something seen in carrier operations generally with a > named SBC in front , keep in mind that FS is a great software but it > doesn't hold against SBC's that use special processors like ASIC to perform > transcoding , sip parsing , tone detection ,etc etc > > > Disclaimer : post written deep into the night... > > On 20/11/13 00:09, Andre wrote: > > Hi, I am testing out my FreeSWITCH 1.2.14 64 bit windows 8.1 on my > Alienware Laptop with a very fast SSD and I7 processor with 8 GB DDR3 Ram. > > > > I have been using a pjsip test tool to test FreeSWITCH to see how it > performs. Tell me if I’m wrong here, I believe if I sent “too many” calls > to FreeSWITCH that it queues up the calls and sooner or later uses all the > memory on my PC then just stops working. I do have the CPS sent to 1500 > for testing but I never hit that high, I hit about 200 CPS. > > > > The CDRs that save to my database show it takes about 8 seconds from start > time to end time. It does start out only taking milliseconds then it > progresses to the 8 second mark. > > > > The calls are just going to FreeSWITCH then I’m sending back a 503. > > > > I have every mod turned off, logging set to 0, Core db going to a SQL > Server and I even set FreeSWITCH to real time. > > > > Can anyone give me advise on what I can do to get better performance? > I’ve also tested it out on my real server but in about 3 minutes of 50 cps > FreeSWITCH uses up 3gb of memory and stops responding. > > > > I’m at a loss here on what else to do. > > Andre > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [The entire original message is not included.]

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Polycom Park and Retrieve
by Spencer Thomason 22 Nov '13

22 Nov '13

While not strictly a FreeSWITCH question, I'm trying to emulate the Polycom park and retrieve behavior in the following video: http://www.youtube.com/watch?v=iNf5IBBqwvY Does anyone have any pointers as to change the label on a line key dynamically? Thanks, Spencer

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Dial plan not working
by Joli Martinez 21 Nov '13

21 Nov '13

I have Kamailio as my front end and I am passing 41 XXX XXX XXXX to FS. I need to be able to make an outbound call. I have added 41_carrier.xml to "/usr/local/freeswitch/conf/dialplan/default" and created carrier.xml file in "/usr/local/freeswitch/conf/sip_profiles/external" when I dial from my softphone 41 XXX XXX XXXX it keeps saying "sofia/internal/" FS does pick up the correct digits but, it never hits the external profile. I have verified that the provider is registered to FS. What am I doing wrong?

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Configure Freeswitch as an IMS application server
by Thomas Titty 21 Nov '13

21 Nov '13

Hello, I installed Freeswitch 1.5.7 and am trying to configure it as an telephony application server in an IMS network. Here is the high-level signaling flow that I am considering. 1. Two SIP users (User-A and User-B) registered with S-CSCF in the IMS network (not with Freeswitch) 2. User-A makes a voice call to User-B. 3. S-CSCF forwards the SIP INVITE to Freeswitch. 4. Freeswitch acting as a B2BUA changes SDP to anchor the media flow and sends SIP INVITE to User-B back to S-CSCF. 5. SIP 200 OK from User-B follows the same path as INVITE in reverse direction. I went through the dial-plan documentation and could not figure out whether such a configuration is possible. Please let me know whether such a configuration is feasible and any broad pointers on how to make it work would be appreciated. Thanks, Thomas

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How to cancel a bridge with multiple groups of destinations
by Ali Pey 21 Nov '13

21 Nov '13

Hello, We need to be able to cancel a bridge through bind_digit_action if the caller decides to go to voicemail. hupall cancels the bridge, but if there are other parallel destinations in the bridge command, it continues calling on those destinations. How can I stop bridge all together with parallel destinations? Regards, Ali

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can't make outbound call using GV
by Richard Cruz 21 Nov '13

21 Nov '13

I can't make outbound call using GV, but i can receive inbound call FS/ GV this is the log: 2013-11-21 00:13:57.533156 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables 2013-11-21 00:13:57.533156 [NOTICE] switch_channel.c:926 New Channel dingaling/gtalk/+16873946400(a)voice.google.com[c936466e-5dc6-4436-a4b6-7f0bebd4f011] 2013-11-21 00:13:57.533156 [DEBUG] mod_dingaling.c:1861 (dingaling/gtalk/+ 16873946400(a)voice.google.com) State Change CS_NEW -> CS_INIT 2013-11-21 00:13:57.533156 [DEBUG] switch_core_session.c:1224 Send signal dingaling/gtalk/+16873946400(a)voice.google.com [BREAK] 2013-11-21 00:13:57.533156 [DEBUG] mod_dingaling.c:1396 dingaling/gtalk/+ 16873946400(a)voice.google.com CHANNEL KILL 2013-11-21 00:13:57.533156 [DEBUG] switch_core_state_machine.c:382 (dingaling/gtalk/+16873946400(a)voice.google.com) Running State Change CS_INIT 2013-11-21 00:13:57.533156 [DEBUG] switch_core_state_machine.c:421 (dingaling/gtalk/+16873946400(a)voice.google.com) State INIT 2013-11-21 00:13:57.533156 [NOTICE] mod_dingaling.c:1158 Ring-Ready dingaling/gtalk/+16873946400(a)voice.google.com! 2013-11-21 00:13:57.533156 [DEBUG] mod_dingaling.c:1111 Don't have my codec yet here's one 2013-11-21 00:13:57.533156 [DEBUG] mod_dingaling.c:1131 Send Describe [PCMU@8000] 2013-11-21 00:13:57.952519 [DEBUG] mod_dingaling.c:3017 using Existing session for 8943672113 2013-11-21 00:13:57.952519 [DEBUG] mod_dingaling.c:1131 Send Describe [PCMU@8000] 2013-11-21 00:13:58.032574 [DEBUG] mod_dingaling.c:3017 using Existing session for 8943672113 2013-11-21 00:13:58.032574 [DEBUG] mod_dingaling.c:1032 Stun Lookup Local 192.168.1.2:24622 2013-11-21 00:13:58.172699 [INFO] mod_dingaling.c:1042 Stun Success 98.219.30.20:24622 2013-11-21 00:13:58.172699 [DEBUG] mod_dingaling.c:1056 Send Candidate 98.219.30.20:24622 [LxTLJwjnAXG9UvkO] 2013-11-21 00:13:58.272796 [DEBUG] mod_dingaling.c:3017 using Existing session for 8943672113 2013-11-21 00:13:58.272796 [DEBUG] mod_dingaling.c:3356 3 candidates 2013-11-21 00:13:58.272796 [DEBUG] mod_dingaling.c:3376 candidate 74.125.139.127:19305 PASS ACL wan.auto 2013-11-21 00:13:58.272796 [DEBUG] mod_dingaling.c:3428 Acceptable Candidate 74.125.139.127:19305 2013-11-21 00:13:58.372880 [DEBUG] mod_dingaling.c:3017 using Existing session for 8943672113 2013-11-21 00:13:58.372880 [DEBUG] mod_dingaling.c:3488 hungup dingaling/gtalk/+16873946400(a)voice.google.com 2013-11-21 00:13:58.372880 [DEBUG] mod_dingaling.c:741 Terminate called from line 3489 state=CS_INIT 2013-11-21 00:13:58.372880 [DEBUG] switch_channel.c:2849 (dingaling/gtalk/+ 16873946400(a)voice.google.com) Callstate Change DOWN -> HANGUP 2013-11-21 00:13:58.372880 [NOTICE] mod_dingaling.c:762 Hangup dingaling/gtalk/+16873946400(a)voice.google.com [CS_INIT] [NORMAL_CLEARING] 2013-11-21 00:13:58.372880 [DEBUG] switch_channel.c:2872 Send signal dingaling/gtalk/+16873946400(a)voice.google.com [KILL] 2013-11-21 00:13:58.372880 [DEBUG] mod_dingaling.c:1396 dingaling/gtalk/+ 16873946400(a)voice.google.com CHANNEL KILL 2013-11-21 00:13:58.372880 [DEBUG] switch_core_session.c:1224 Send signal dingaling/gtalk/+16873946400(a)voice.google.com [BREAK] 2013-11-21 00:13:58.372880 [DEBUG] mod_dingaling.c:1396 dingaling/gtalk/+ 16873946400(a)voice.google.com CHANNEL KILL 2013-11-21 00:13:58.372880 [DEBUG] mod_dingaling.c:3490 End Call

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collecting stats on RTMP calls
by Jon Hancock 21 Nov '13

21 Nov '13

Hi, I'm using FreeSWITCH 1.5.x as config'd in BigBlueButton 0.81. I need to find a way to measure useful stats on latency, dropped connects/packets. I have no current need to worry about scale. I'm only trying to find settings for decent voice quality. I've installed voipmonitor (w/o web gui) and so far haven't figured out how to get it to capture the call data. It looks like voipmonitor is running, the mysql db has the schema but no data when I'm running a call. Is there some FreeSWITCH module in need to get it work with voipmonitor? Any ideas appreciated. thanks, Jon

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configuration params for webrtc
by Vincent Xia 21 Nov '13

21 Nov '13

i was trying to test jssip + freeswitch 1.2.14 for webrtc, it's said that freeswitch 1.2.14 supports webrtc and the only thing to do is adding in sip_profiles\internal.xml the following:      which i suppose is already in internal.xml of freeswitch 1.2.14 package, but it's not, the internal.xml looks very same as the 1.2.10 one. i downloaded the .bz2 package instead of doing a git clone, is this the issue?

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