February 2019 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Zoiper goes crazy with registrations RPORT bug?
by Rajil Saraswat 28 Feb '19

28 Feb '19

Hello, I have a Zoiper android client with Freeswitch server version 1.8.4-5-749a6e108b~64bit. If i select RPORT option in Zoiper along with TLS/TCP transport protocol, i get hundreds for registration requests per minute. Zoiper customer support thinks that Freeswitch is at fault here. The message i received from them is below. Is there any fix for this? ------------ The issue in your case is that the topmost Via header does not have the mandatory received parameter. According to RFC 3581 (An Extension to the Session Initiation Protocol (SIP) for Symmetric Response Routing) Section 4. Server Behavior: "When a server compliant to this specification (which can be a proxy or UAS) receives a request, it examines the topmost Via header field value. If this Via header field value contains an "rport" parameter with no value, it MUST set the value of the parameter to the source port of the request. This is analogous to the way in which a server will insert the "received" parameter into the topmost Via header field value. * In fact, the server MUST insert a "received" parameter* containing the source IP address that the request came from, even if it is identical to the value of the "sent-by" component. Note that this processing takes place independent of the transport protocol." This is a server side issue and it is related most likely to the version of FreeSWITCH that you are using. ---------- Thanks

2 4

Modify SIP timers (T1X64) to improve failover
by Marc S 27 Feb '19

27 Feb '19

Hello, I'm testing gateways failover with several gateway for bridge AND ping setting. It works but INVITE timeout and SIP Options timeout is quite long (32 seconds) because of timer-T1X64 setting (it's what i hav readen) I would like to modify timer-T1X64, for example : 3000. Invite failover would and SIP option timout would now be : 3 seconds. Tests are OK, Invite failover and gateway marked down is very quick. My question : is there any impact on behavior of other transactions in SIP protocol ? Is there no risk to reduce timer-T1X64 from 32000 to 3000 ? Thanks, Best regards, Marc

1 0

Codec negotiation with g729.
by Mansi Gupta 26 Feb '19

26 Feb '19

Hello! I am trying to make a call using codec g729,but repeatedly i was getting the error of codec negotiation.However,i have resolved it. Although it offers a list of codec with different payloads for the same codec but it does not match with other audio codec.I have to match the audio codec g729 with the other one. Here are few relevant lines of the log: Local SDP: v=0 o=FreeSWITCH 1551144396 1551144397 IN IP4 192.168.1.123 s=FreeSWITCH c=IN IP4 192.168.1.123 t=0 0 m=audio 22412 RTP/AVP 0 18 102 9 8 101 104 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:102 opus/48000/2 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G729:18:8000:20:8000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMU:0:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G729:18:8000:20:8000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[opus:116:48000:20:0:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[G722:9:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [GSM:3:8000:20:13200:1]/[PCMA:8:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[PCMU:0:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[G729:18:8000:20:8000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[opus:116:48000:20:0:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[G722:9:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [speex:110:8000:20:0:1]/[PCMA:8:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4504 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:98:8000:30:0:1]/[PCMU:0:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:98:8000:30:0:1]/[G729:18:8000:20:8000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:98:8000:30:0:1]/[opus:116:48000:20:0:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:98:8000:30:0:1]/[G722:9:8000:20:64000:1] 2019-02-26 02:40:10.077544 [DEBUG] switch_core_media.c:4449 Audio Codec Compare [iLBC:98:8000:30:0:1]/[PCMA:8:8000:20:64000:1] Can anyone please point,how to resolve it? Thanks. -- Sent from: http://freeswitch-users.2379917.n2.nabble.com/

3 2

Re: [Freeswitch-users] where g729 fs-latest-installer script now ?
by Social Boh 26 Feb '19

26 Feb '19

So you don't update MPL 1.1 with MPL. 2.0 because the 1.1 is not compatible with GPL while 2.0 is compatible with GPL. Regards --- I'm SoCIaL, MayBe El 19/02/2019 a las 14:53, Ken Rice escribió: > > I was pretty clear on the my statements about taking funding away from > the FreeSWITCH Project and the devs by doing such thing > > Also the bcg module is GPLv2 license violation waiting to be noticed… > you cant combine MPL and GPL code and have a resulting MPL codebase. > Its not allowed by the GPL. So you you are also ripping off the GPL > coder. > > Now you are polluting a MPL code base with the GPL violating the > Copyright holder of the MPL’s software rights to determine how his > code is licensed. > > This happens all the time. People think oh this code is free, and yes > most of the code is free, but the devs still have mortgages, children > to feed, computers and related hardware to keep updated so that they > can continue to support that hardware that everyone is using > currently. Website hosting for something like FreeSWITCH isnt free. It > requires over 1/2 a rack of colo’d equipment to keep the website, > mailing list, build servers, git repos, wiki etc etc etc going. > > $10 for a channel is a pretty cheap way to support that seeing as that > most of the time you don’t even need G729 > > *From:*Social Boh <social(a)bohboh.info> > *Sent:* Tuesday, February 19, 2019 1:03 PM > *To:* Ken Rice <krice(a)freeswitch.org> > *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer > script now ? > > Can you explain me why I can't use the mod_bcg729 with FreeSWITCH? > > Thank you > > --- > I'm SoCIaL, MayBe > > El 19/02/2019 a las 13:04, Ken Rice escribió: > > My statement wasn’t to you, but to the people that continually > want to pirate someone elses G729 code and refuse to support the > projects developers. > > Its one of those things that grinds my gears. > > K > > *From:* FreeSWITCH-users > <freeswitch-users-bounces(a)lists.freeswitch.org> > <mailto:freeswitch-users-bounces@lists.freeswitch.org> *On Behalf > Of *Yuriy Nasida > *Sent:* Tuesday, February 19, 2019 6:47 AM > *To:* António Silva <asilva(a)wirelessmundi.com> > <mailto:asilva@wirelessmundi.com> > *Cc:* freeswitch-users(a)lists.freeswitch.org > <mailto:freeswitch-users@lists.freeswitch.org> > *Subject:* Re: [Freeswitch-users] where g729 fs-latest-installer > script now ? > > Ken, not sure what you talking about. I already bought a lot of > g729 licenses for my servers and I use them a lot of years. But > now I can't use them normally at Deb 9 + fs1.8 + new g729 install > script because of issues I described. > > Thanks a lot Antonio! I will try this way. > > On Tue, 19 Feb 2019 at 11:41, António Silva > <asilva(a)wirelessmundi.com <mailto:asilva@wirelessmundi.com>> wrote: > > try using the .deb and installed then manuall.: > > Get the packages from here: > http://files.freeswitch.org/repo/deb/freeswitch-1.8/pool/main/f/ > > Installed them manually: > > dpkg -x > freeswitch-mod-com-g729_1.8.5~7~14d59c97e0~jessie_amd64.deb / > dpkg -x > freeswitch-license-server_0.0.1~42~731842be7d~jessie_amd64.deb / > > I don't know if the installer will remain, from the jira > https://freeswitch.org/jira/browse/FS-11641 i got the > impression that now the solution is to use the debian packages. > > On 13/02/2019 18:10, Yuriy Nasida wrote: > > I have installed freeswitch-license-server using > fs-201902072050-installer and got issue below. > > FS doesn't listen port 8021 and I can't connect to it and > run any command using fs_cli. > > All works fine in case I kill process > freeswitch-license-server. > > Can anybody comment please all these very weird things > with commercial g729 at FS ? > > Thanks. > > On Wed, 13 Feb 2019 at 13:36, Yuriy Nasida > <ynasida(a)gmail.com <mailto:ynasida@gmail.com>> wrote: > > Well. Somebody updated > http://files.freeswitch.org/g729/ and now it > has fs-latest-installer. Thanks a lot whoever you are!!! > > But... Now it has not install instructions :) > > On Wed, 13 Feb 2019 at 12:25, Yuriy Nasida > <ynasida(a)gmail.com <mailto:ynasida@gmail.com>> wrote: > > Um... Nobody uses g729 licences anymore ? :) > > On Tue, 12 Feb 2019 at 20:07, Yuriy Nasida > <ynasida(a)gmail.com <mailto:ynasida@gmail.com>> wrote: > > Hi guys, > > Does anybody know why site > http://files.freeswitch.org/g729/ contain only > install instrunction now ? > Where fs-latest-installer-v1.6 script or > something like this ? > > Please advice. > > _________________________________________________________________________ > > Professional FreeSWITCH Services > > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > > https://freeswitch.com > > > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://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 > > https://freeswitch.com > > -- > > Saludos / Regards / Cumprimentos > > António Silva > > > > _________________________________________________________________________ > > Professional FreeSWITCH Services > > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > > https://freeswitch.com > > Official FreeSWITCH Sites > > https://freeswitch.com/oss > > https://freeswitch.org/confluence > > https://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 > > https://freeswitch.com >

5 5

Change SIP Contact header on b-leg
by Кирилл Преслер 22 Feb '19

22 Feb '19

Provider not accepting calls because of "bad" Contact header sip:gw+ss@10.50.244.15:5060;transport=udp;gw=ss ITSP needs sip:username@10.50.244.15:5060, where username should be passed from dialplan for every call and there is no params. I've tried following variables with set and export, but nothing seems to work. sip_contact_user sip_outgoing_contact_uri sip_invite_contact_params

3 2

1.8 DTLS calls
by Jurijs Ivolga 21 Feb '19

21 Feb '19

Hi, I'm trying to migrate from 1.6.20 to 1.8.5, but I'm facing a problem with DTLS calls(media_webrtc=true). Does anybody has same issues as me? I have following setup: RTP=>Freeswitch=>DTLS In 1.8.5, when Freeswitch receive 200 from DTLS endpoint, but it never sends it back to RTP endpoint. Here some snippet from logs, where call stuck: 2019-02-20 09:50:07.368784 [INFO] switch_rtp.c:3832 Changing audio DTLS state from OFF to HANDSHAKE 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8815 sofia/internal/ 9999999999999999999(a)sip.myapp.net Set 2833 dtmf send payload to 101 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8822 sofia/internal/ 9999999999999999999(a)sip.myapp.net Set 2833 dtmf receive payload to 101 2019-02-20 09:50:07.368784 [DEBUG] switch_core_media.c:8845 sofia/internal/ 9999999999999999999(a)sip.myapp.net Set rtp dtmf delay to 40 In 1.6.20 everything works as expected: 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7180 sofia/internal/ 9999999999999999999(a)sip.myapp.net Set 2833 dtmf send payload to 101 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7187 sofia/internal/ 9999999999999999999(a)sip.myapp.net Set 2833 dtmf receive payload to 101 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:7210 sofia/internal/ 9999999999999999999(a)sip.myapp.net Set rtp dtmf delay to 40 2019-02-20 11:12:40.129989 [NOTICE] sofia.c:8218 Channel [sofia/internal/ 9999999999999999999(a)sip.myapp.net] has been answered 2019-02-20 11:12:40.129989 [DEBUG] switch_channel.c:3773 (sofia/internal/ 9999999999999999999(a)sip.myapp.net) Callstate Change RINGING -> ACTIVE 2019-02-20 11:12:40.129989 [DEBUG] switch_core_codec.c:248 sofia/internal/ 1234567890(a)sip.myapp.net Restore previous codec PCMA:8. 2019-02-20 11:12:40.129989 [DEBUG] switch_core_media.c:6861 Audio params are unchanged for sofia/internal/1234567890(a)sip.myapp.net. 2019-02-20 11:12:40.129989 [DEBUG] mod_sofia.c:850 Local SDP sofia/internal/ 1234567890(a)sip.myapp.net: Jurijs

2 6

Connecting endpoint to domain within Free Switch
by Koketso Mabuse 21 Feb '19

21 Feb '19

Hi there Scenario: Fresh installed FusionPBX: * Default domain with local IP, * created a 2nd domain as per tutorial, * created extensions. o tried to register extensions on 2nd domain failure + changed domain on extion to point to 1st main domain, end point registers. Issue: how do I create 2 extensions eg: 1000 for domain 1 and 1000 for domain 2 and register them both on main domain? Domain:mtcc exmple: Username: 1000 Password: pass123 host: 192.168.1.130 Domain: tluka Username: 1000 Password: pass123 host: 192.168.1.130

2 3

setting contact domain in dialplan not working
by Roman Dissauer 20 Feb '19

20 Feb '19

Is there a way to get the FS public IP replaced by a custom domain name in contact header using the dialplan? tried: <action application="export" data="sip_contact_host=test.domain.tld“/> ...not working <action application=„set" data="sip_contact_host=test.domain.tld“/> ...not working <action application="export" data="sip_invite_domain=test.domain.tld“/> ...not working <action application=„set" data="sip_invite_domain=test.domain.tld“/> ...not working <action application="export" data=„sip_contact_user=bla(a)test.domain.tld"/> ...not working <action application=„set" data=„sip_invite_domain=bla(a)test.domain.xn--tld/-db7a> ...not working only thing working is setting the parameter in gateway config, <param name="contact-host" value=„test.domain.tld“/> but this is not working for me because I need to do it dynamically in dialplan! Any hint for me? thank you! Roman

2 3

[FreeSWITCH Install Issue]
by Bilal Abbasi 20 Feb '19

20 Feb '19

Hi Users, I am having issues with installing FreeSWITCH 1.8.5 on Debian 9.6, Here is the error i am getting this error when i run make command. CC src/libfreeswitch_la-switch_msrp.lo CC src/libfreeswitch_la-switch_vad.lo CC src/libfreeswitch_la-switch_vpx.lo CXX src/switch_cpp.lo CXXLD libfreeswitch.la *** Warning: Linking the shared library libfreeswitch.la against the *** static library libs/libvpx/libvpx.a is not portable! ar: `u' modifier ignored since `D' is the default (see `U') Making all in . make[2]: Entering directory '/usr/src/freeswitch-1.8.5' CC src/freeswitch-switch.o CCLD freeswitch /usr/bin/ld: warning: libssl.so.1.0.2, needed by //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1 /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1 CC src/tone2wav-tone2wav.o CCLD tone2wav /usr/bin/ld: warning: libssl.so.1.0.2, needed by //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1 /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1 CC src/fs_encode-fs_encode.o CCLD fs_encode /usr/bin/ld: warning: libssl.so.1.0.2, needed by //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libssl.so.1.1 /usr/bin/ld: warning: libcrypto.so.1.0.2, needed by //usr/lib/x86_64-linux-gnu/libcurl.so.4, may conflict with libcrypto.so.1.1 make[2]: Leaving directory '/usr/src/freeswitch-1.8.5' Making all in src make[2]: Entering directory '/usr/src/freeswitch-1.8.5/src' Making all in mod make[3]: Entering directory '/usr/src/freeswitch-1.8.5/src/mod' Unknown target Makefile:721: recipe for target '-all' failed make[3]: *** [-all] Error 1 make[3]: Leaving directory '/usr/src/freeswitch-1.8.5/src/mod' Makefile:591: recipe for target 'all-recursive' failed make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory '/usr/src/freeswitch-1.8.5/src' Makefile:3494: recipe for target 'all-recursive' failed make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory '/usr/src/freeswitch-1.8.5' Makefile:1255: recipe for target 'all' failed make: *** [all] Error 2 Google did not helped me much on this, can somebody point me the right direction. Regards Abbasi

4 5

Re: [Freeswitch-users] VoIP encryption recommendations
by Joel Serrano 20 Feb '19

20 Feb '19

Bipin, Did you try the linphone tunnel? AFAIR they encrypt SIP+RTP on client (requires linphone obviously), they also provide a server which receives such connections and pass over the unencrypted SIP+RTP to the backend. I think it's worth the try... some years ago it got around most blocks we tested. We ended up not implementing it but the initial tests did look good, don't know nowadays though... On Tue, Feb 19, 2019 at 11:59 AM Bipin Patel via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: Bipin Patel <bipin(a)xbipin.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 23:58:21 +0400 > Subject: Re: [Freeswitch-users] VoIP encryption recommendations > > Btw we have tried almost all codecs and currently using g711u but it's > just impossible to get through easily not to mention the isp even > blacklists whole data center subnets if they find any data center to be a > safe heaven for VoIP providers. > > They even have this mechanism that home users if attempt to connect to any > blocked service then their IP ends up on a blacklist and then the filtering > gets worse for them until they reboot the router which gets them a new IP > and things get back to normal > > No wonder this region is considered a million dollar market where > Microsoft and Facebook itself can't manage to keep Skype and WhatsApp calls > even running no matter what. > > > On February 19, 2019 11:50:45 PM Bipin Patel <bipin(a)xbipin.com> wrote: > >> Hi, >> >> >> The whole sip protocol is blocked and udp VPN don't connect and tcp ones >> they delay packets a lot so calls end up heavily choppy. On mobile data the >> restrictions are even heavier and if packets are anywhere close to VoIP or >> VoIP over VPN etc they get filtered. Etisalat is the isp which buys >> blocking equipment from some vendor in UK who specialize in blocking VoIP >> and VPN. Last I was told by some person working there was they use a lot of >> L7 packet inspectors. >> >> Secondly it's not about setting up custom solutions for any company or >> client but we generate a lot of retail traffic so users need something that >> they can run on mobile etc like a customized dialer. Untill now I used to >> give them a openvpn profile which they used to run and then use Zoiper to >> place calls but all that is blocked now. >> >> Webrtc seems to work as of now coz it's new but there isn't a webrtc >> based mobile dialer till now which anyone can install and just use it to >> place calls >> >> >> >> On February 19, 2019 9:47:46 PM Sergey Safarov <s.safarov(a)gmail.com> >> wrote: >> >>> In your case need >>> 1) increase ptime value to 40; >>> 2) use codec without compression; >>> 3) need to enable VAD feature; >>> >>> Also you can route torrent client network traffic via VPN together with >>> VoIP traffic. This will mask VoIP packets and not allow math your traffic >>> to VoIP profile on ISP equipment. >>> >>> >>> вт, 19 февр. 2019 г. в 19:53, Bipin Patel via FreeSWITCH-users < >>> freeswitch-users(a)lists.freeswitch.org>: >>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Bipin Patel <bipin(a)xbipin.com> >>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>> Cc: >>>> Bcc: >>>> Date: Tue, 19 Feb 2019 20:24:20 +0400 >>>> Subject: VoIP encryption recommendations >>>> hi, >>>> >>>> i would like the ask the community about VoIP encryption, currently in >>>> few middle east countries VoIP is officially blocked. The isp are so >>>> aggressive that they use all sorts of fancy tools to block it including >>>> skype calls, whatsapp calls etc and are very successful in doing it. So far >>>> companies like voipswitch and recently few others have been providing >>>> tunneling mechanisms to get over this but recently UDP traffic is heavily >>>> filtered and they go to the extreme of checking packet length and pattern >>>> and artificially introduce delay, jitter or simply block it if the number >>>> of hits are high. Switching to TLS/SRTP also doesnt help, it works with >>>> some isp but as soon as you try same using mobile data it stops working coz >>>> they match packet length and block based on the profile. ZRTP doesnt work >>>> coz a normal RTP streams needs to start and then it starts encrypting it >>>> but those initial RTP get blocked. >>>> >>>> With lack of any more VoIP encryption protocols its almost getting >>>> impossible to bypass block so has anyone have any ideas of any other modern >>>> form of encryption which can be used for VoIP (btw VPN are also blocked and >>>> more over if packet size increases then nothing works on mobile data). >>>> >>>> The market demand of skype replacements is also extremely high coz >>>> skype, hangouts, whatsapp video, instagram video, viber etc etc, u name it >>>> and its blocked. >>>> >>>> -- >>>> Regards, >>>> Bipin >>>> >>>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Bipin Patel via FreeSWITCH-users < >>>> freeswitch-users(a)lists.freeswitch.org> >>>> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> >>>> Cc: >>>> Bcc: >>>> Date: Tue, 19 Feb 2019 08:53:29 -0800 (PST) >>>> Subject: [Freeswitch-users] VoIP encryption recommendations >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Services >>>> sales(a)freeswitch.com >>>> https://freeswitch.com >>>> >>>> Official FreeSWITCH Sites >>>> https://freeswitch.com/oss >>>> https://freeswitch.org/confluence >>>> https://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 >>>> https://freeswitch.com >>> >>> >> > > > > ---------- Forwarded message ---------- > From: Bipin Patel via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 19 Feb 2019 11:59:01 -0800 (PST) > Subject: Re: [Freeswitch-users] VoIP encryption recommendations > _________________________________________________________________________ > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com

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