opalvoip-user — Discussions relating to OpalVoip user questions

Re: [Opalvoip-user] GNU Gatekeer 5.2 released

[S-trace <s-...@li...>]

[Opalvoip-user] GNU Gatekeer 5.2 released

[Jan W. <ja...@wi...>]

Hi,

I have just released GNU Gatekeeper version 5.2.

You can download it from https://www.gnugk.org/h323download.html

This release that has a rewritten networking implementation (aka
"large-fdset") that allows GnuGk to scale to higher numbers of calls
per server than previous versions.

This new implementation replaces the old hack to extend the select()
system call by using poll() which enables GnuGk to handle huge numbers
of sockets at the same time. The new implementation also works on
Windows, but has been tested mostly on the different Unix versions we
support.

Please note that the relevant configure option when comping GnuGk
source code has changed to --enable-large-fdset. There is no need to
specify a maximum number of sockets any more.

This release also has a few bug fixes, eg. for using LUA scripts with
shared libraries and for memory leaks in the error handling of H.235
password authentication.


Whats new ?

- re-implement LARGE_FDSET using poll(), enable with configure
  option --enable-large-fdset
- ExternalIP is automatically added to the default domains
- support running LUA scripts that require dynamic libraries
- change default for [TLS] CipherList= to allow elliptical curve ciphers
- BUGFIX(gkauth.h) fix memory leak in H.235 password auth
- BUGFIX(gkacct.cxx) set known, but unavailable accounting parameters
  to empty string
- BUGFIX(ProxyChannel.cxx) fix setting UDP source IP on Windows when
  compiled for Vista or higher


Enjoy!

--
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

[Opalvoip-user] GNU Gatekeer 5.1 is out

[Jan W. <ja...@wi...>]

Hi,

I have released GNU Gatekeeper version 5.1.

You can download it from https://www.gnugk.org/h323download.html

The main new feature in this release is H.245 multiplexing.
Together with the long supported RTP multiplexing it allows GnuGk to
handle a large amount of calls from H.460 endpoints using just 5
ports total.

Whats new ?

- support for H.245 multiplexing with H.460.18:
  [RoutedMode] EnableH245Multiplexing=1, H245MultiplexPort=1722
- improved interop with Lifesize Icon (H.235), Scopia VC240 (H.460.18)
  and Yealink Mobile (H.460.19)
- improved detection of neighbor gatekeeper availability
- public IP detection for Google Cloud
- new feature to let GnuGk send an event if port detection fails

There were also a number of bug fixes, please see changes.txt or the
blog post:
https://blog.gnugk.org/2019/01/gnu-gatekeeper-5-1.html

Enjoy!

--
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

[Opalvoip-user] Problem to build minimal 3.16.2

[Alain B. <ala...@po...>]

Hello,

Even Ubuntu 18.04 provides only version 3.10.10. That’s really a pitty !
So I tried to build libPt 2.16.2  and libOpal 3.16.2 from sources.
No problem for libPt but build fails for libOpal.
To do it, I read this page, which takes version 3.4.2 as reference !?! 
Maybe it’s the reason why I get the following errors :

[CXX] src/codec/opalwavfile.cxx
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx: In instantiation of ‘static void PWAVFilePluginFactory<Factory, Instance>::Register(const typename Factory::Key_T&, const PWAVFilePluginValidFormat&) [with Factory = PFactory<PWAVFileFormat, unsigned int>; Instance = PWAVFileFormatPlugin; typename Factory::Key_T = unsigned int]’:
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:428:81:   required from here
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:413:29: error: no matching function for call to ‘PFactory<PWAVFileFormat, unsigned int>::Register(const Key_T&, PWAVFilePluginFactory<PFactory<PWAVFileFormat, unsigned int>, PWAVFileFormatPlugin>*&, bool)’
In file included from /usr/local/include/ptclib/pwavfile.h:46:0,
                 from /home/alainb/opal-3.16.2/include/codec/opalwavfile.h:40,
                 from /home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:35:
/usr/local/include/ptlib/pfactory.h:387:5: note: candidate: static bool PFactory<AbstractClass, KeyType>::Register(const Key_T&, PFactory<AbstractClass, KeyType>::WorkerBase_T*) [with AbstractClass = PWAVFileFormat; KeyType = unsigned int; PFactory<AbstractClass, KeyType>::Key_T = unsigned int; PFactory<AbstractClass, KeyType>::WorkerBase_T = PFactoryTemplate<PWAVFileFormat, const unsigned int&, unsigned int>::WorkerBase]
/usr/local/include/ptlib/pfactory.h:387:5: note:   candidate expects 2 arguments, 3 provided
/usr/local/include/ptlib/pfactory.h:387:5: note: candidate: static bool PFactory<AbstractClass, KeyType>::Register(const Key_T&, PFactory<AbstractClass, KeyType>::Abstract_T*, bool) [with AbstractClass = PWAVFileFormat; KeyType = unsigned int; PFactory<AbstractClass, KeyType>::Key_T = unsigned int; PFactory<AbstractClass, KeyType>::Abstract_T = PWAVFileFormat]
/usr/local/include/ptlib/pfactory.h:387:5: note:   no known conversion for argument 2 from ‘PWAVFilePluginFactory<PFactory<PWAVFileFormat, unsigned int>, PWAVFileFormatPlugin>*’ to ‘PFactory<PWAVFileFormat, unsigned int>::Abstract_T* {aka PWAVFileFormat*}’
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx: In instantiation of ‘static void PWAVFilePluginFactory<Factory, Instance>::Register(const typename Factory::Key_T&, const PWAVFilePluginValidFormat&) [with Factory = PFactory<PWAVFileFormat, PCaselessString>; Instance = PWAVFileFormatPlugin; typename Factory::Key_T = PCaselessString]’:
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:429:85:   required from here
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:413:29: error: no matching function for call to ‘PFactory<PWAVFileFormat, PCaselessString>::Register(const Key_T&, PWAVFilePluginFactory<PFactory<PWAVFileFormat, PCaselessString>, PWAVFileFormatPlugin>*&, bool)’
In file included from /usr/local/include/ptclib/pwavfile.h:46:0,
                 from /home/alainb/opal-3.16.2/include/codec/opalwavfile.h:40,
                 from /home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:35:
/usr/local/include/ptlib/pfactory.h:387:5: note: candidate: static bool PFactory<AbstractClass, KeyType>::Register(const Key_T&, PFactory<AbstractClass, KeyType>::WorkerBase_T*) [with AbstractClass = PWAVFileFormat; KeyType = PCaselessString; PFactory<AbstractClass, KeyType>::Key_T = PCaselessString; PFactory<AbstractClass, KeyType>::WorkerBase_T = PFactoryTemplate<PWAVFileFormat, const PCaselessString&, PCaselessString>::WorkerBase]
/usr/local/include/ptlib/pfactory.h:387:5: note:   candidate expects 2 arguments, 3 provided
/usr/local/include/ptlib/pfactory.h:387:5: note: candidate: static bool PFactory<AbstractClass, KeyType>::Register(const Key_T&, PFactory<AbstractClass, KeyType>::Abstract_T*, bool) [with AbstractClass = PWAVFileFormat; KeyType = PCaselessString; PFactory<AbstractClass, KeyType>::Key_T = PCaselessString; PFactory<AbstractClass, KeyType>::Abstract_T = PWAVFileFormat]
/usr/local/include/ptlib/pfactory.h:387:5: note:   no known conversion for argument 2 from ‘PWAVFilePluginFactory<PFactory<PWAVFileFormat, PCaselessString>, PWAVFileFormatPlugin>*’ to ‘PFactory<PWAVFileFormat, PCaselessString>::Abstract_T* {aka PWAVFileFormat*}’
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx: In instantiation of ‘static void PWAVFilePluginFactory<Factory, Instance>::Register(const typename Factory::Key_T&, const PWAVFilePluginValidFormat&) [with Factory = PFactory<PWAVFileConverter, unsigned int>; Instance = PWAVFileConverterPlugin; typename Factory::Key_T = unsigned int]’:
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:432:81:   required from here
/home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:413:29: error: no matching function for call to ‘PFactory<PWAVFileConverter, unsigned int>::Register(const Key_T&, PWAVFilePluginFactory<PFactory<PWAVFileConverter, unsigned int>, PWAVFileConverterPlugin>*&, bool)’
In file included from /usr/local/include/ptclib/pwavfile.h:46:0,
                 from /home/alainb/opal-3.16.2/include/codec/opalwavfile.h:40,
                 from /home/alainb/opal-3.16.2/src/codec/opalwavfile.cxx:35:
/usr/local/include/ptlib/pfactory.h:387:5: note: candidate: static bool PFactory<AbstractClass, KeyType>::Register(const Key_T&, PFactory<AbstractClass, KeyType>::WorkerBase_T*) [with AbstractClass = PWAVFileConverter; KeyType = unsigned int; PFactory<AbstractClass, KeyType>::Key_T = unsigned int; PFactory<AbstractClass, KeyType>::WorkerBase_T = PFactoryTemplate<PWAVFileConverter, const unsigned int&, unsigned int>::WorkerBase]
/usr/local/include/ptlib/pfactory.h:387:5: note:   candidate expects 2 arguments, 3 provided
/usr/local/include/ptlib/pfactory.h:387:5: note: candidate: static bool PFactory<AbstractClass, KeyType>::Register(const Key_T&, PFactory<AbstractClass, KeyType>::Abstract_T*, bool) [with AbstractClass = PWAVFileConverter; KeyType = unsigned int; PFactory<AbstractClass, KeyType>::Key_T = unsigned int; PFactory<AbstractClass, KeyType>::Abstract_T = PWAVFileConverter]
/usr/local/include/ptlib/pfactory.h:387:5: note:   no known conversion for argument 2 from ‘PWAVFilePluginFactory<PFactory<PWAVFileConverter, unsigned int>, PWAVFileConverterPlugin>*’ to ‘PFactory<PWAVFileConverter, unsigned int>::Abstract_T* {aka PWAVFileConverter*}’
cc1plus: warning: unrecognized command line option ‘-Wno-unused-private-field’
cc1plus: warning: unrecognized command line option ‘-Wno-unused-private-field’
/usr/local/share/ptlib/make/post.mak:138: recipe for target '/home/alainb/opal-3.16.2/lib_linux_x86_64/obj/opalwavfile.o' failed
make[2]: *** [/home/alainb/opal-3.16.2/lib_linux_x86_64/obj/opalwavfile.o] Error 1
/usr/local/share/ptlib/make/post.mak:115: recipe for target 'optshared' failed
make[1]: *** [optshared] Error 2
make[1] : on quitte le répertoire « /home/alainb/opal-3.16.2 »
/usr/local/share/ptlib/make/autoconf.mak:171: recipe for target 'build_top_level' failed
make: *** [build_top_level] Error 2

Thanks for your help,

Alain

⁣

Alain BONNEFOY
19, rue Pierre Méchain
26000 Valence
(0)4 75 80 39 10 
Site Internet : ponant-technologies.com

Envoyé par TypeApp ​

[Opalvoip-user] GNU Gatekeer 5.0 released

[Jan W. <ja...@wi...>]

Hi,

I'm happy to announce the release of GNU Gatekeeper version 5.0.

This version has new features and a few bug fixes. You can download it
from https://www.gnugk.org/h323download.html

Whats new ?

- support for Azure and Alibaba Cloud in addition to AWS
- performance optimizations, especially for multiplexed RTP and LUA
- compatible with OpenSSL 1.1.x
- switch to translate Facility transfers into gatekeeper TCS0 reroutes

There were also a number of bug fixes, please see changes.txt for
details.

Enjoy!

-- 
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

[Opalvoip-user] GNU Gatekeer 4.9 released

[Jan W. <ja...@wi...>]

Hi,

I have just released GNU Gatekeeper version 4.9. This version has
new features and a few bug fixes. You can download it from
https://www.gnugk.org/h323download.html

Whats new ?

We have 2 new accounting modules: HtttpAcct and AMQPAcct that allow
you to send accounting events via HTTP GET or POST to a web service
or push them into a RabbitMQ queue.

There are also many new accounting placeholders that you can use with
any of the accounting modules and there is a new accounting event
'reject' to track calls rejected with ARJ that went unnoticed before.

The new RTP inactivity checking allows you to drop calls if there
wasn't any RTP activity for a defined amount of time.

GeoIP authentication has been significantly updated to support all
RAS and all Q.931 messages and to support the new Maxmind database
format (GeoIP2).

There were also a few bug fixes:
- BUGFIX(ProxyChannel.cxx) fix crash while handling RTP packets
- BUGFIX(RasTbl.cxx) fix disconnecting unregistered endpoints
- BUGFIX(RasSrv.cxx) fix crash in some Avaya endpoints when receiving
GCF with a gatekeeperIdentifier
- BUGFIX(RasSrv.cxx) fix crash when using IPv6
- BUGFIX(ProxyChannel.cxx) fix handling of CloseLogicalChannel

Enjoy!

-- 
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

[Opalvoip-user] GNU Gatekeeper 4.8 released

[Jan W. <ja...@wi...>]

Hi,

I have just released GNU Gatekeeper version 4.8. This version has many
new features. You can download the new version from
https://www.gnugk.org/h323download.html

Overview:

New maintenance mode: When you need to take down your GnuGk server
(eg. for an OS update), you can switch GnuGk to maintenance mode where
it will only allow ongoing calls to finish and automatically redirects
all idle endpoints to an alternate GnuGk server.
The status port command is "MaintenanceMode <alternate IP>".

Detailed information about ongoing calls: You can now display lots of
information about each ongoing call (codecs, bandwidth used, IPs etc.).
The web interface has been extended to to show this information.
See https://www.gnugk.org/images/web7.jpg

Easier installation on AWS and inside docker containers. You can now
let GnuGk automatically detect the public IP of your AWS server, even
from within a docker container. You can also automatically insert your
public/external IP into your trace file names to store logs from many
servers in the same directory.

Extended API: Call routing with external applications has been
expanded. You can now set the display names for participants and
desired reject codes on the status port. You can also access the
vendor information of all registered endpoints. The web interface has
been extended to provide this information, too.

HttpPasswordAuth has been greatly extended to fetch password
information from backend servers. We now use curl to support https
and you can add many new placeholders in your queries.

Extended screening and rewriting of display names and calling party
names.

Important bug fixes: Multiplexed RTP is now much more robust and
password authentication to parent gatekeepers has been fixed. There
are also interop fixes for TCP keep-alives.


Please see the full change log below for more details.

Changes from 4.7 to 4.8
=======================
- HttpPasswordAuth: support https and add new placeholders
- PrintAllRegistrationsVerbose now also shows the endpoint vendor
- new status port command: MaintenanceMode
- new status port command: PrintCallInfo
- allow placeholder %{gkip} and %{external-ip} in [LogFile] Filename=
- fetch AWS public/elastic IP if ExternalIP=AWSPublicIP
- new commandline switch: -e / --externalip
- extend status port command RouteReject to set reject reason
- extend status port commands RouteToAlias, RouteToGateway etc. to set
display IE for calling and called
- new switch: [LogFile] DeleteOnRotation=1 to delete the old logfile
when rotating instead of renaming it
- new switches: [RoutedMode] AppendToCallingPartyNumberIE= /
PrependToCallingPartyNumberIE= to add any string before or after the
calling party number IE when ScreenCallingPartyNumberIE=RegisteredAlias
- when [RoutedMode] ScreenCallingPartyNumberIE= is set to
RegisteredAlias, GnuGk sets calling party number IE to the registered
alias (forced screening)
- delete DisplayIE when [RoutedMode] ScreenDisplayIE=Delete
- new switch [Endpoint] Authenicators=
- new default: [RoutedMode] GnuGkTcpKeepAliveMethodH225=EmptyFacility
- new default: [RoutedMode] H460KeepAliveMethodH225=EmptyFacility for
Cisco interop
- new setting "None" for keep-alive methods

- BUGFIX(ProxyChannel.cxx) fix bugs in H.460.19 RTP multiplexing
- BUGFIX(ProxyChannel.cxx) don't send H.460 keep-alive to non-H.460
  party when calling H.460 party
- BUGFIX(Routing.cxx) show called port in RouteRequests (as documented)
- BUGFIX(GkClient.*) fix password authentication with parent
- BUGFIX(Routing.cxx) remove semicolon and pipe chars from vendor
  string in RouteRequests
- better handling of IPv6 GRQ without RAS address
- BUGFIX(ProxyChannel.cxx) turn off encryption proxy if DH key is
  negotiated, but TCS doesn't contain any H.235 entries
- BUGFIX(ProxyChannel.cxx) fix running in proxy mode on FreeBSD when
  one Home IP is set
- BUGFIX(ProxyChannel.cxx) fix DisableSettingUDPSourceIP=1 for Windows,
  NetBSD, OpenBSD and Solaris
- BUGFIX(yasocket.cxx) fix LARGE_FDSET for NetBSD, OpenBSD and Solaris


-- 
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

[Opalvoip-user] Console and simple samples

[Ali Al-N. <113...@st...>]

Hello robert,

I am using opal-3.16.2

I was trying to make a call using console sample between two parties (peer
to peer  with gnu gatekeeper) through h323 i set those arguments (
--rec-aud-dev -- play-aud-dev --no-sip )... it worked fine !


But if i want it to be a secure rtp session should i add --h323-crypto
"type of encryption" ? Do i have to set same type of encryption on both
parties otherwise the call will be aborted ?

When i try to make an h323 call on both  parties after assigning a crypto
suite  the call only lasts for about 20 seconds then i loose the sound  but
not the call ...


Sip call works fine !
But when i try to set sip crypto "AES_CM_128_HMAC_SHA1_32 " on both parties
, it says call failed because  it could not find common media capabilities
!


What should i do !?

Thanks in advance

[Opalvoip-user] Console and simple samples

[Ali Al-N. <113...@st...>]

I am Using opal-3.16.2 ..

When i try to make an h323 call on both  samples the call only lasts for
about 20 seconds then i loose the sound ...

Sip call works fine ! But when i try to srt sip crypto
"AES_CM_128_HMAC_SHA1_32 or 80 " on both parties then it says call failed
because  it could not find common media capabilities !

What should i do !?


Thanks in advance

[Opalvoip-user] Console opal-3.16.2

[Ali Al-N. <113...@st...>]

Hello opal-voip team
Hello robert,

I am sending this again and i hope you answer me.

I was trying to make a call using console sample between two parties
through h323 i set those arguments ( --rec-aud-dev -- play-aud-dev --no-sip
)... it worked fine !

But if i want it to be a secure rtp session should i add --h323-crypto
"type of encryption" ? Do i have to set same type of encryption on both
parties otherwise the call will be aborted ?

By setting this argument do i guarantee having message integrety and
authentication as well as encryption of the packets ?
And is the key true random generated ?  And then it is exchanged in a
secure way ?

Thanks in advance
Best regards

On Dec 23, 2017 8:00 PM, "Ali Al-Najjar" <113...@st...>
wrote:

> Hello opal-voip team,
>
> I was trying to make a call using console sample between two parties
> through h323 i set those arguments ( --rec-aud-dev -- play-aud-dev --no-sip
> )... it worked fine !
>
> But if i want it to be a secure rtp session should i add --h323-crypto
> "type of encryption" ? Do i have to set same type of encryption on both
> parties otherwise the call will be aborted ?
>
> By setting this argument do i guarantee having message integrety and
> authentication as well as encryption of the packets ?
> And is the key true random generated ?  And then it is exchanged in a
> secure way ?
>
> Thanks in advance
> Best regards
>
>

[Opalvoip-user] Console opal-3.16.2

[Ali Al-N. <113...@st...>]

Hello opal-voip team,

I was trying to make a call using console sample between two parties
through h323 i set those arguments ( --rec-aud-dev -- play-aud-dev --no-sip
)... it worked fine !

But if i want it to be a secure rtp session should i add --h323-crypto
"type of encryption" ? Do i have to set same type of encryption on both
parties otherwise the call will be aborted ?

By setting this argument do i guarantee having message integrety and
authentication as well as encryption of the packets ?
And is the key true random generated ?  And then it is exchanged in a
secure way ?

Thanks in advance
Best regards

Re: [Opalvoip-user] SRTP AND H323s

[Robert J. <ro...@vo...>]

> On 18 Dec 2017, at 4:52 pm, Ali Al-Najjar <113...@st...> wrote:
> 
> Hello opalvoip team , 
> 
> I am using opal library v13

If you mean 3.13, then that is far too old.

> I was trying to use the simple opal sample and there is an option about using h323s.
> 
> What is meant by h323s ( secure h323 ) ?

The signalling channel is over TLS. Nothing to do with media. Similar to "sips".

> What kind of security does it deliver ? 

It hides things like phone numbers that are pin the signalling channel. It isn't very common.

> Is it an aes encryption of the data ? 

Not of the above h323s, no. That is TLS. For H.235 media, yes, it is.

> is srtp enabled in this sample (simple opal )?

Yes. If it was compiled in.

> And how could i check if srtp is enabled in any sample ?? 

It will tell you.

> what are the steps for enabling srtp from scratch ?? 

Use the latest version for a start. You will need to have OpenSSL available in your system, other than that, It should just compile and automatically negotiate.


----------
Robert Jongbloed
Vox Lucida Pty. Ltd.

[Opalvoip-user] SRTP AND H323s

[Ali Al-N. <113...@st...>]

Hello opalvoip team ,

I am using opal library v13
I was trying to use the simple opal sample and there is an option about
using h323s.

What is meant by h323s ( secure h323 ) ?
What kind of security does it deliver ?
Is it an aes encryption of the data ?

And one more question :

is srtp enabled in this sample (simple opal )?
And how could i check if srtp is enabled in any sample ??
what are the steps for enabling srtp from scratch ??

Thanks in advance
Best regards

Re: [Opalvoip-user] Question concerning SRTP

[Robert J. <ro...@vo...>]

Search to source for SetMediaCryptoSuites() function, and how it is used in sample code.


----------
Robert Jongbloed
Vox Lucida Pty. Ltd.




On 16 Dec 2017, at 2:40 pm, Ali Al-Najjar <113...@st... <mailto:113...@st...> > wrote:

Hello opalvoip team,

I am using ur library and it is great .
I want to ask you a question about SRTP.
How could i enable and disable srtp on h323 ?

I hope you would reply soon...

Thanks in advance
Best regards
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[Opalvoip-user] Question concerning SRTP

[Ali Al-N. <113...@st...>]

Hello opalvoip team,

I am using ur library and it is great .
I want to ask you a question about SRTP.
How could i enable and disable srtp on h323 ?

I hope you would reply soon...

Thanks in advance
Best regards

[Opalvoip-user] Fwd: Compilation error in Opal Voip

[Abhilash K.V <abh...@gm...>]

Hello,

I am new to Opal Voip , trying to compile this using MAC machine and want
to run OpenPhone app on my iPhone and Android devices.

Could you pls help me on this.

1. From where to clone the required components
2. Steps for building it with dependency information

I followed
http://wiki.opalvoip.org/index.php?n=Main.BuildingPTLibMacOSX

But when I compiled OpenPhone getting following Errors.

Maro undefined CODEC_FLAG_EMU_EDGE, CODEC_FLAG_TRUNCATED ...


Thanks,
Abhilash.

Re: [Opalvoip-user] [PTLIB] Can PString.GetSize() return negative values?

[Robert J. <ro...@vo...>]

It is never negative.

On many platforms PINDEX is defined as size_t which is typically unsigned.

----------
Robert Jongbloed
Vox Lucida Pty. Ltd.




On 20 Nov 2017, at 6:22 pm, Soul Trace via Opalvoip-user <opa...@li... <mailto:opa...@li...> > wrote:

Hello


I have simple question: Can PString.GetSize() return negative values?

Looked into documentation, but found only that it returns PINDEX (aka int) value.

But is it safe to compare it with unsigned value?


Thank you.


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[Opalvoip-user] [PTLIB] Can PString.GetSize() return negative values?

[Soul T. <S-...@li...>]

Hello


I have simple question: Can PString.GetSize() return negative values?

Looked into documentation, but found only that it returns PINDEX (aka 
int) value.

But is it safe to compare it with unsigned value?


Thank you.

[Opalvoip-user] GNU Gatekeeper 4.7 has been released (security update)

[Jan W. <ja...@wi...>]

Hi,

GNU Gatekeeper version 4.7 has just been released.

This version is purely a security update and has no new features. All
users are encouraged to update, especially if you use port detection
(IgnoreSignaledIPs=1) you should update ASAP.

It has been discovered that GnuGk is vulnerable in some configurations
for RTP bleed attacks (https://rtpbleed.com/). By updating to version
4.7 only the first packets in each media stream influence the media
destination.

To further secure your configuration, you can set

[Proxy]
RestrictRTPSources=Net

to only accept RTP from the same class C network that the call
signaling came from. Please beware that this may break a few valid calls
where this condition isn't met.

You can download the new version from
https://www.gnugk.org/h323download.html


Please see the full change log below.

Changes from 4.6 to 4.7
=======================
- fixes for RTP Bleed
- new switch [Proxy] RestrictRTPSources=IP or Net to limit accepting RTP
  from the call signal IPs or the respective class C network
- new switch [Proxy] LegacyPortDetection=1 to keep port detection help
  for some very old and broken endpoints that will make your gatekeeper
  vulnerable to RTP Bleed attacks
- BUGFIX(ProxyChannel.cxx) replace @ip or ip## from aliases when using
  RedirectCallsToGkIP
- BUGFIX(ProxyChannel.cxx) better initialization of sendmsg() structs
- new command line option: now you can use -S instead of --strict (needed
  on BSD systems)

-- 
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

[Opalvoip-user] GNU Gatekeeper 4.6 is out

[Jan W. <ja...@wi...>]

Hi,

I'm happy to announce that GNU Gatekeeper version 4.6 has just been
released.

This version has a few new features as well as bug fixes.

New features:
- least used routing: distribute calls evenly between gateways or MCUs
  (new switch [RasSrv::ARQFeatures] LeastUsedRouting=1)
- ability to log to the Unix syslog instead of the trace file
  (new switch [LogFile] TraceToSyslog=1)
- new authentication module TwoAliasAuth
  this is not very safe, but you can use it with endpoints that do not
  support any password transmission
- new switch [CTI::MakeCall] Bandwidth= to set the maximum bandwidth
  for the calls generated by the GnuGk status port API
- status port command: UnregisterEP <ep-id>
- a number of switches to fine tune TCP keepalives
- new switch to remove load balancers from the call path
  ([RoutedMode] RedirectCallsToGkIP=1)

Bug fixes:
- fixed TCP keepalive for H.460 calls
- fixes to port detection for unregistered calls
- audio fix when GnuGk adds encryption to calls
- many smaller fixes

You can download the new version from
https://www.gnugk.org/h323download.html


Please see the full change log below.

Changes from 4.5 to 4.6
=======================
- new switch: [RoutedMode] RedirectCallsToGkIP=1
- new switches: [RoutedMode] H460KeepAliveMethodH225=, H460KeepAliveMethodH245=,
  GnuGkTcpKeepAliveMethodH225=, GnuGkTcpKeepAliveMethodH245=
- BUGFIX(ProxyChannel.cxx) TCP keep-alives for H.460.18 calls weren't always
  enabled correctly
- don't open a status port listener if [Gatekeeper::Main] StatusPort=0
- BUGFIX(Toolkit.cxx) remove trailing chars before checking for DefaultDomain
- add callID to H.245 trace messages for easier debugging
- BUGFIX(ProxyChannel.cxx) forward ReleaseComplete from remaining party while
  doing call reroute
- BUGFIX(ProxyChannel.cxx) drop un-en/decryptable RTP packets at end of call
  when adding encryption
- new status port command: UnregisterEP <ep-id>
- BUGFIX(RasSrv.cxx) remove IPv6 addresses before processing RRQs when IPv6 is not enabled
- send Facility message as as non-H.460.18 keep-alive for H.225
- send non-standard H.245 userIndication as non-H.460.18 keep-alive for H.245
- new switch [RoutedMode] DisableGnuGkH245TcpKeepAlive=1
- new switch [LogFile] TraceToSyslog=1 to send trace output to syslog (Unix only)
- BUGFIX(ProxyChannel.cxx) fix port detection for re-opened channels with IgnoreSignaledIPs=1
- new switch [CTI::MakeCall] Bandwidth= to set the maximum bandwidth for the call
- new switch [RasSrv::ARQFeatures] LeastUsedRouting=1 to select the least used gateway
- new authentication module TwoAliasAuth


-- 
Jan Willamowius, Founder of the GNU Gatekeeper Project
EMail  : ja...@wi...
Website: https://www.gnugk.org
Support: https://www.willamowius.com/gnugk-support.html

Relaxed Communications GmbH
Frahmredder 91
22393 Hamburg
Geschäftsführer: Jan Willamowius
HRB 125261 (Amtsgericht Hamburg)
USt-IdNr: DE286003584

Re: [Opalvoip-user] one way video

[Robert J. <ro...@vo...>]

Hussein,
  simpleopal has been deprecated for many years now. I must assume you are using a very, very old version. Please try the latest STABLE version.
 
----------
Robert Jongbloed
Vox Lucida Pty. Ltd.




On 31 May 2017, at 1:29 pm, Sergei Nikulov <ser...@gm... <mailto:ser...@gm...> > wrote:

Hello,

2017-05-31 12:11 GMT+03:00 hussein bazzi <hus...@ou... <mailto:hus...@ou...> >:
hello

please i need your help in simpleopal

when using h323 mode , the video screen appear only on the caller part ,

while nothing appear on the other part , and when the caller is terminated ,
i get this message : libv4l2 error dequeuing buf invalid arguments ..

how can i solve it ?


Unfortunately, I'm unable to help you.

Could you please use the opal-users mailing list address for your questions?

-- 
Best Regards,
Sergei Nikulov

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Re: [Opalvoip-user] one way video

[Sergei N. <ser...@gm...>]

Hello,

2017-05-31 12:11 GMT+03:00 hussein bazzi <hus...@ou...>:
> hello
>
> please i need your help in simpleopal
>
> when using h323 mode , the video screen appear only on the caller part ,
> while nothing appear on the other part , and when the caller is terminated ,
> i get this message : libv4l2 error dequeuing buf invalid arguments ..
>
> how can i solve it ?
>

Unfortunately, I'm unable to help you.

Could you please use the opal-users mailing list address for your questions?

-- 
Best Regards,
Sergei Nikulov

Re: [Opalvoip-user] video call

[Sergei N. <ser...@gm...>]

2017-05-30 10:46 GMT+03:00 hussein bazzi <hus...@ou...>:
> hi
>
> i am having a problem with videocall
>
> i am using opal 3.10.10

Could you please read the previous message from Robert?

The summary - that version is several years old and a lot has happened
in that time.

>
> the video call is established in sip mode , but not in h323 mode .
>
> what could be the reason ?
>
>
> ------------------------------------------------------------------------------
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>



-- 
Best Regards,
Sergei Nikulov

[Opalvoip-user] video call

[hussein b. <hus...@ou...>]

hi

i am having a problem with videocall

i am using opal 3.10.10

the video call is established in sip mode , but not in h323 mode .

what could be the reason ?

Re: [Opalvoip-user] A call from sip-in with PCMA to sip-out-with-PCMA failed, indicating a weird media negociation error

[<Jam...@gm...>]

  Hi, Robert,

  Will do and let you know the result. Thanks



Jam...@gm...
 
From: Robert Jongbloed
Date: 2017-05-29 03:53
To: Jam...@gm...
CC: opalvoip-user
Subject: Re: [Opalvoip-user] A call from sip-in with PCMA to sip-out-with-PCMA failed, indicating a weird media negociation error
James,
  version 3.10.10 ? Really? That is several years old and a lot has happened in that time. Three stable release in fact. I would strongly recommend using current stable version, 3.16.

  
----------
Robert Jongbloed
Vox Lucida Pty. Ltd.




On 28 May 2017, at 2:47 pm, Jam...@gm... <jam...@gm...> wrote:

   Sorry, All,
   
   I attached two documents over 40k. Having it zipped, and post it again. Hope you all could read them now.
   
   James Su

From: Jam...@gm...
Date: 2017-05-28 23:39
To: opalvoip-user
Subject: A call from sip-in with PCMA to sip-out-with-PCMA failed, indicating a weird media negociation error
   Hi, opalvoip-user,

   Just trying to make a opalvoip application to implement a very simple softswitch using Opalvoip 3.10.10 as below,
   - With a sip call coming in, the application just modifying the called number in the call and forwarding it out via a sip trunk.
   - Codec is very simple as PCMA for A party and B party. There is no requirement for transcoding.
   
   Now the issue I met is that when the called subscriber answers the call, the application would start releasing the call, as below
   - An incoming call from subscriber SIPA to my application 
   - My application forwards the call to subcriber SIPB
   - SIPB alerting
   - SIPA hearing alert sound
   - SIPB answers
   - my application starts to release the call once receiving the 200Ok from subscriber SIPB
   
    I did some work in the tracing log as attachment and found that there is a media negociation error happening. But as you know, actually there is no need for media negociation, just PCMA in and PCMA out, which you should be able to find in the log. Any thought please kindly let me know. Thanks. 
   
   BTW, Please refer to the attachment for the tracing log and my main application implementation. 
   
   Thanks a lot. 

   James Su
<log_and_main_implement.rar>------------------------------------------------------------------------------
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