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Re: [Freeswitch-users] softswitch
by Dan Christian Bogos 02 May '14

02 May '14

Hi Hassan, Yes, I know the details of the configuration (done the benchmarks myself). Actually we are pretty serious with these benchmarks, by default CGRateS coming with it's own stress tool (cgr-tester binary) which you can use to benchmark your own rating profiles. The machine we have used is a virtual host with a one Intel 3.4 GHz CPU and 1GB of RAM. Hardisk does not matter since we do not store anything for calculations on HDD. All the data we use in rating calculations is cached using own cache mechanism. On this machine we got the following stats (querying cost for one emulated call repeatedly -100k times) : 1. No network involved in query, (querying internally within the same process): 163655 requests per second. Applicability: CGR-SM component directly attached to FreeSWITCH with internal connection to the rater. 2. Remote RPC using GOB as transport: 12501 requests per second. Applicability: CGR-SM running on the same host with FreeSWITCH, remote rater (eg: charging for a cluster of FreeSWITCHes). 3. Remote RPC using JSON-RPC as transport (worst case scenario): 6221 requests per second. Applicability: Third party written session manager using CGR-Rater component for it's rating and implementing it's own call handling logic. I should mention that the number of requests per second should not decrease with the number of destinations you search since we use individual hashes in our cache and tested so far with 6.5 Mil. destinations without impact on performance. Happy to go deeper in your questions, not sure if appropriate to abuse here the mailing list for them, otherwise you can always join our mailing list or irc channel for a more interactive conversation. DanB PS: For the sake of correctness the screenshoot of the performance benchmarks I am talking in this email is available to be directly sent to you if interested (not accepted by the mailing list because of size). On 02.05.2014 13:43, freeswitch-users-request(a)lists.freeswitch.org wrote: > CGRateS seems to be a very interesting project! > > The features page says "a machine of modest configuration" was used to > benchmark up to 13,000 req/sec. Would you happen to know details of this > "modest configuration"? > > Regards > HASSAN

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Video-only call
by Richard Screene 02 May '14

02 May '14

Hello, Does FreeSWITCH support receiving calls that contain only a video component? (I am trying to use FreeSWITCH to distribute WebRTC screenshare streams to users. It works fine with audio and video) The INVITE looks like: INVITE sip:1273936153_290033102@a.b.c.d SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356 max-forwards: 69 To: <sip:1273936153_290033102@a.b.c.d> From: "Rich" <sip:rscreene@gmail.com>;tag=mgu8fejfq8 call-id: llc4a6rbqb7886s43nhl CSeq: 7243 INVITE Contact: <sip:127.0.0.1:7071;transport=tcp> allow: ACK,CANCEL,BYE,OPTIONS,INVITE,MESSAGE supported: outbound user-agent: JsSIP 0.4.0-devel content-length: 1824 Content-Type: application/sdp v=0 o=- 4235676450597780712 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE video a=msid-semantic: WMS 1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd m=video 34116 RTP/SAVPF 100 116 117 c=IN IP4 e.f.g.h a=rtcp:34116 IN IP4 e.f.g.h a=candidate:2668419503 1 udp 2122260223 192.168.47.203 56323 typ host generation 0 a=candidate:2668419503 2 udp 2122260223 192.168.47.203 56323 typ host generation 0 a=candidate:3515819359 1 tcp 1518280447 192.168.47.203 0 typ host generation 0 a=candidate:3515819359 2 tcp 1518280447 192.168.47.203 0 typ host generation 0 a=candidate:1401022012 1 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0 a=candidate:1401022012 2 udp 1686052607 w.x.y.z 7940 typ srflx raddr 192.168.47.203 rport 56323 generation 0 a=candidate:3671437516 1 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0 a=candidate:3671437516 2 udp 41885439 e.f.g.h 34116 typ relay raddr w.x.y.z rport 7940 generation 0 a=ice-ufrag:vD/mqbgKKKvb+vwg a=ice-pwd:xbg5CyTfAIQMCD3JTmjU9IGw a=ice-options:google-ice a=fingerprint:sha-256 E8:6D:CF:75:43:C6:D9:C6:90:8E:FD:15:CD:FC:86:A2:90:E6:A3:BE:CF:A1:12:64:60:2F:26:89:96:C0:16:18 a=setup:actpass a=mid:video a=extmap:2 urn:ietf:params:rtp-hdrext:toffset a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=sendrecv a=rtcp-mux a=rtpmap:100 VP8/90000 a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=rtcp-fb:100 goog-remb a=rtpmap:116 red/90000 a=rtpmap:117 ulpfec/90000 a=ssrc:2554093369 cname:L3M//yZhkStQCiOE a=ssrc:2554093369 msid:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd 908bd517-4202-4348-a36d-f474090f3cc2 a=ssrc:2554093369 mslabel:1dVXakoYx2u585L7G09f2C45EW8vP8c1XVcd a=ssrc:2554093369 label:908bd517-4202-4348-a36d-f474090f3cc2 In the log I see the following message: 2014-05-02 13:35:43.267313 [DEBUG] switch_core_media.c:2223 No audio codec available And FreeSWITCH responds with: SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 0.0.0.0:7071;branch=z9hG4bK801683;rport=7071;received=a.b.c.d Via: SIP/2.0/TCP 127.0.0.1:7071;branch=z9hG4bKllc4a6rbqb7886s43nhl Via: SIP/2.0/TCP 1q582tjutlhh.invalid;branch=z9hG4bK7896356 Max-Forwards: 692014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (sofia/external/rscreene(a)gmail.com) State REPORTING From: "Rich" <sip:rscreene@gmail.com>;tag=mgu8fejfq8 2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:102 sofia/external/rscreene(a)gmail.com Standard REPORTING, cause: INCOMPATIBLE_DESTINATION To: <sip:1273936153_290033102@a.b.c.d>;tag=H3e25S58vjNZQ call-id: llc4a6rbqb7886s43nhl 2014-05-02 13:35:43.267313 [DEBUG] switch_core_state_machine.c:818 (sofia/external/rscreene(a)gmail.com) State REPORTING going to sleep CSeq: 7243 INVITE User-Agent: Drum 1.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "1273936153_290033102" <sip:1273936153_290033102@a.b.c.d>;party=calling;privacy=off;screen=no Do I have to configure something to allow video-only calls? Or, is it a limitation of FreeSWITCH? If it is a limitation of FreeSWITCH is it easy to remove? Many thanks, Richard Richard Screene Senior Developer Drum Web Meetings +44 1273 936125 www.thisisdrum.com Drum is the collaboration solution by NetDev Ltd. Registered in England and Wales Company Number 04741258

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Re: [Freeswitch-users] softswitch
by Dan Christian Bogos 02 May '14

02 May '14

Hi Joli, We are using (and developing) CGRateS [1]. It is opensource, real-time, actively maintained and quite feature-rich. Let me know if any questions ;). DanB [1] http://cgrates.org/ On 02.05.2014 04:47, freeswitch-users-request(a)lists.freeswitch.org wrote: > ------------------------------ > > Message: 3 > Date: Thu, 1 May 2014 21:24:09 -0400 > From: Joli Martinez<mrjoli021(a)gmail.com> > Subject: [Freeswitch-users] softswitch > To: FreeSWITCH Users Help<freeswitch-users(a)lists.freeswitch.org> > Message-ID:<513284CE-8510-42E3-BFAA-3DBEE27CBC27(a)gmail.com> > Content-Type: text/plain; charset=us-ascii > > Hello, > > I am creating a softswitch to resell minutes. I will be using Kamailio and Freeswitch. I am looking for a real-time billing engine and accounting. I would prefer an open-source solution but open to suggestions. What would you recommend? > > Thanks,

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softswitch
by Joli Martinez 02 May '14

02 May '14

Hello, I am creating a softswitch to resell minutes. I will be using Kamailio and Freeswitch. I am looking for a real-time billing engine and accounting. I would prefer an open-source solution but open to suggestions. What would you recommend? Thanks,

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Friday Bughunt and ClueCon Weekly! Announcements
by Ken Rice 01 May '14

01 May '14

Greetings FreeSWITCHers! Yesterday on CC Weekly, we had Travis Cross demo a nice tool for using git bisect to help track down bugs in FreeSWITCH! If you missed it torrents of the call (and previous calls) are up at http://torrents.freeswitch.org/ Tomorrow at 2PM Easter Join us on 888 for The FreeSWITCH Friday Bug Hunt! Have an open Jira you want to discuss or get some help with? Just want to help us work through some issues? Maybe you have a problem and you arent sure its a bug or a config issue. Join us on the bridge tomorrow! Access to the bridge is sip:888@conference.freeswitch.org or visit http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions for additional access methods. Next Week (May 7) on ClueCon Weekly TJ Vanderpoel from Geoforce, Inc will be joining us with a tmux demo and how to. Tmux is one of those sysadmin tools that can help make your life easier if you live in terminal sessions all day! Keep in mind the calls start at 1PM Eastern, but the bridge is open early for meet and greet before and after the call! Join us early so you don¹t miss anything! Coming up on ClueCon Weekly May 14th Giovanni Maruzzelli aka gmaruzz on IRC will be updating us on GSMOpen. How to use GSM Mobile devices with FreeSWITCH. May 21st Lee Howard will be stopping by to update us on what¹s going on in the Hylafax+ world. May 28th Eric Crist, Director of IT with ClaimLynx, Inc and regular FreeSWITCH user and contributor will stop by to talk about Faxing with FreeSWITCH. If you have a topic you would like to have presented on ClueCon Weekly, Drop me an email (here or offlist) and lets get it on the schedule! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Voice channels in recorded file is out of sync
by Jayesh Nambiar 30 Apr '14

30 Apr '14

Hi, This problem has been reported several times in JIRA as well as lists without a proper solution. The problem is that the A-Leg runs a little ahead than the B-Leg in the recording only. The actual conversation is clear. I have following inputs to it: 1) This happens when the calls are made from a specific network and for calls that go beyond 20 minutes. 2) I tried this on Ubuntu 12.04 as well as Ubuntu 13.10, the problem still remains the same. 3) I've tried this with multiple version of freeswitch and the latest one I tested with is the following version: Version 1.5.12b+git~20140419T062924Z~f408dae2af~64bit (git f408dae 2014-04-19 06:29:24Z 64bit). In all versions the problem is consistent. 4) I tried with suppress-cng=true as well as false but the problem is still there. 5) I've added parameters like "bridge_generate_comfort_noise=true" before bridge. 6) I've tried with "record_waste_resources=true" before calling record_session. 7) I've tried with "record_fill_cng=true" before calling record_session. None of these helped me in solving this problem. I understand that it is something in the network probably, but I'll be greatful if anyone can point me for what to check on the network!! When can freeswitch record_session introduce such problem so that I can start looking into those directions. The endpoint being used was Grandstream GXP-1400. Is there a specific condition under which the freeswitch recording channels can go out of sync?? Because it starts well but suddenly when the conversation reaches 15-20 minutes the overlapping starts; meaning the channels start going out of sync !! Any help or pointers will be of great help. Thanks, --- Jayesh

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Adding IVR menus
by Sean Devoy 30 Apr '14

30 Apr '14

Hi, I have added a new IVR menu in conf/autoload_configs/ivr.conf.xml but I keep getting unknown menu errors. Reloadxml does not seem to help. Do I have to restart FS to load new IVR menus? Sean

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Am I the only one?
by Francis 30 Apr '14

30 Apr '14

I was having trouble with starting freeswitch the usual way: ~# service freeswitch start because of an error in one of the configs. So I decided to run it manually from the command line so I could see the error output. And guess what? That causes the databases to be owned by root! Of course, this means that freeswitch will ALWAYS fail when starting as a service because it can't access the database. I only confirmed this after doing: ~# su - freeswitch -s /bin/bash and starting freeswitch again, which presented the tell-tale read only errors that I was suspecting. It also started a race condition,with freeswitch endlessly trying (and failing) to write to core.db. I also discovered that using php-ssh to access files does the same thing, after using Pydio to view logs and suddenly they were owned by root as well. The solution I've gone with is: 1. Give ownership and SGID of all web-accessible files to www-data 2. Add freeswitch to the www-data group 3. Always 'su - freeswitch' if I want to run freeswitch directly to see errors from the shell I'm not sure if that's the best way, but it's working well so far. I'm also finding Pydio and Gate One are a great combination for a purely web managed freeswitch install. Francis

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Terminating ZRTP call over two FS's bridge.
by Assaf Dahary 30 Apr '14

30 Apr '14

Hi, I have two FS's (FS1, FS2) with ZRTP enable in two sites. I can make end-to-end ZRTP calls between two CSipSimple/ZRTP clients over the two FS's bridge where both FS's are set to 'proxy_media=true'. CSipSimple -> FS1 ---Internet -à FS2 -> CSipSimple. Now I connected a PBX to each FS with SIP-Trunk. PBX -> FS1 ------- FS2 -> PBX. When calling locally from CSipSimple to a PBX extension on the same FS then the ZRTP is terminated by the FS as MITM which is OK (FS dialplan 'proxy_media=false'). What I cannot figure out is how to terminate ZRTP call between CSipSimple/FS1 and FS2/PBX.ext and vice versa. CSipSimple -> FS1 ------- FS2 -> PBX.ext. I've tried to setup: FS1 with 'proxy_media=true' (passthrough ZRTP) + FS2 with 'proxy_media=false' (terminate ZRTP). OR FS1 and FS2 with 'proxy_media=false' . All results with ' ZRTP not negotiated on both sides; disabling ZRTP passthru mode'. Any help will be appreciated. Regards Assaf

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Inisdeout vs Vanilla config
by Francis 30 Apr '14

30 Apr '14

I'm just curious about the difference between the two - in terms of functionality and purpose. Are the configs all the same, or are some altered? Cheers!

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