Re: [SR-Users] Doing automatic unregister when a WEBSOCKET connection is closed.
On 5/13/13 6:44 AM, Juha Heinanen wrote: Peter Dunkley writes: I suppose an alternative function that just needs ruid could be written - but it would be much less efficient as it would have to linearly search all records (unless an additional hash on ruid is added - and a DB index on it too). thanks for the explanation. now that we have ruid, the rest of usrloc has not clearly kept up with it. a new hash on ruid should be added. usrloc table already has unique index on ruid, which means that in db only mode, it should be quite easy to make the query only based on ruid. For db only the sql query can be done only on ruid (for update and delete), there is a parameter for that now. The aor is needed for memory, because the records are indexed in order to be able to find on save() and lookup() where the aor is in the sip message. Adding a second hash table just to index on ruid will make things more complex from point of view of memory use and synchronization between the two hashes. I said in another email that actually the aorhash is needed, but if aor is known, computing aorhash is trivial. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
Hello, I am not that familiar to troubleshoot asterisk configuration files, but from logs I could see the resulting URI is: INVITEsip:106@(null) SIP/2.0 That is wrong, meaning something incorrect is done when setting it in asterisk. Maybe someone else can help more with asterisk. Cheers, Daniel On 5/13/13 4:02 AM, zhengyw wrote: hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem? ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04 Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt -- View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-x-and-Asterisk-10-7-0-Realtime-Integration-tp118248p118319.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio 4.0.1 and nokia e72 udp
Hello, On 5/12/13 9:48 PM, hiro wrote: I'm using rtpproxy with symmetric NAT, so that is no option. If the packet with offset IP is not received by rtpproxy the call is fine even with symmetric NAT. In case that wasn't clear earlier: I did narrow it down to this single test before posting to simplify and thus make it more clear to everyone. RTP itself is working fine with the E72, only rtpproxy gets confused by that single seemingly useless packet. it looks like rtpproxy needs to be patched, so that learning mode is kept for few packets and it will use the last src port for sending back traffic. Other option could be playing with the firewall rules, to drop always first packet coming on rtpproxy ports range. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] (no subject)
Hello, kamailio does not start, because you have many log errors, like: 0(2935) ERROR: core [modparam.c:163]: set_mod_param_regex: No module matching nathelper|registrar found 0(2935) : core [cfg.y:3570]: parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 479, column 65: Can't set module parameter Go through the log and fix this errors -- the text should be quite explanatory and you get also the line in the config where the error is located. Cheers, Daniel On 5/12/13 9:50 AM, 李启明 wrote: hi, thank you for your help.i am not good at kamailio technology , i execute /etc/init.d/kamailio restart in debug mode,and the console.txt is in attachment.what is wrong? 2013/5/10 Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com Hello, On 5/8/13 2:41 PM, 李启明 wrote: hi, i am from china.i have install a kamailio server with wss support ,and i set 4443 as the port(set 8080 as the ws port);the result of netstat -tnl is: Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State tcp0 0 127.0.0.1:53 http://127.0.0.1:53 0.0.0.0:* LISTEN tcp0 0 127.0.0.1:631 http://127.0.0.1:631 0.0.0.0:* LISTEN tcp0 0 my_ip:44430.0.0.0:* LISTEN tcp0 0 my_ip:50600.0.0.0:* LISTEN tcp0 0 my_ip:50610.0.0.0:* LISTEN tcp0 0 127.0.0.1:3306 http://127.0.0.1:3306 0.0.0.0:* LISTEN tcp0 0 my_ip:80800.0.0.0:* LISTEN but,when i use ws,it's ok;when i use wss,it come up a 106 error!why? run kamailio in debug mode (debug=3 and log_stderror=yes) and watch the logs in the console, you may get hits about what happens. Be sure that the port for wss is on tls. Cheers, Daniel -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 *http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Can I use Kamailio as B2BUA
Hello,
there is a limit in the operating system of the ports that can be used -
a port is short int value, 2^16.
If it is for testing purposes, perhaps you can create many virtual
machines, with a load balancers in front of it.
Anyhow, it might be easier to patch kamailio not to reuse the
connection, look at tcp*.{c,h} files.
Cheers,
Daniel
On 5/11/13 6:27 AM, Kamal Palei wrote:
Hi Daniel
Thanks for reply.
Just curious to know, how many number of calls we can achieve using
the solution you have recommended.
Also the solution, you have recommended, I am not very clear on that
how to do it. Kindly can you explain in detail.
Best Regards
Kamal
On Fri, May 10, 2013 at 12:52 PM, Daniel-Constantin Mierla
mico...@gmail.com mailto:mico...@gmail.com wrote:
Hello,
how many active calls do you expect to have?
At this moment, kamailio does not create local sockets
dynamically, they have to be specified in the configuration.
But if the number of active calls is not big, then you can create
as many sockets as expected calls and then use force send socket
to select on. You can use htable to keep the relation between a
call and a local socket.
Cheers,
Daniel
On 5/10/13 5:45 AM, Kamal Palei wrote:
Dear Kamailio experts
I have a typical use case where I want Kamailio to behave as a
B2BUA.
What I mean here is (assume Kamailio is using TCP for SIP call
establishment)
1. For each call it should create a separate TCP connection
with next proxy in path.
2. When call ends, it should close that connection. If that
call is active for 10hrs, then connection should stay alive
till 10hrs.
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] I need you help-----about Kamailio 3.3.x and Asterisk 10.7.0 Realtime Integration
On 13 May 2013 03:02, zhengyw zhen...@neusoft.com wrote: hello daniel: thank you very much! but I can't find the problem in the asterisk. attachment is asterisk's configure file, kamailio's configure file and data.can you help with this problem? Hi Your video1_sipregs table does not seem correct. Your registered users should have ipaddr and port fields populated. I notice you are missing a few fields which could be a problem. mysql select * from video1_sipregs; ++--+-++--+--+---++ | id | name | fullcontact | ipaddr | port | username | regserver | regseconds | ++--+-++--+--+---++ | 4 | 106 | sip:106@10.11.2.47:5060 ||0 | 106 | | 1368169282 | | 5 | 107 | sip:107@10.11.2.47:5060 ||0 | 107 | | 1368176017 | | 6 | 108 | ||0 | | NULL | 0 | ++--+-++--+--+---++ The sipregs table should have this structure for Asterisk 10.7 as far as I know. CREATE TABLE `sipregs` ( `id` INT(11) NOT NULL AUTO_INCREMENT, `name` VARCHAR(80) NOT NULL DEFAULT '', `fullcontact` VARCHAR(80) NOT NULL DEFAULT '', `ipaddr` VARCHAR(45) DEFAULT NULL, `port` mediumint(5) UNSIGNED NOT NULL DEFAULT '0', `username` VARCHAR(80) NOT NULL DEFAULT '', `regserver` VARCHAR(100) DEFAULT NULL, `regseconds` INT(11) NOT NULL DEFAULT '0', `defaultuser` VARCHAR(80) NOT NULL DEFAULT '', `useragent` VARCHAR(20) DEFAULT NULL, `lastms` INT(11) DEFAULT NULL, PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`) ); -Barry ps:asterisk version 10.70, kamailio version 3.3.1, ubuntu version 12.04 Best Regards, zhengyw kamailio.cfg http://sip-router.1086192.n5.nabble.com/file/n118319/kamailio.cfg sip.conf http://sip-router.1086192.n5.nabble.com/file/n118319/sip.conf extconfig.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extconfig.conf extensions.conf http://sip-router.1086192.n5.nabble.com/file/n118319/extensions.conf db_result.txt http://sip-router.1086192.n5.nabble.com/file/n118319/db_result.txt -- View this message in context: http://sip-router.1086192.n5.nabble.com/I-need-you-help-about-Kamailio-3-3-x-and-Asterisk-10-7-0-Realtime-Integration-tp118248p118319.html Sent from the Users mailing list archive at Nabble.com. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
On 13/05/13 08:56, Daniel Pocock wrote:
For TURN, has anybody tried the TURN server project from Google code?
It appears more advanced than the existing two TURN servers in Debian
(e.g. it has database-backed authentication)
http://code.google.com/p/rfc5766-turn-server/
I have used this TURN Server. I have made some RPMs of it too (.spec
and .patch (patch contains init.d scripts etc)) are attached.
The TURN Server itself seems to work well, though I think there are some
bugs relating to configuration file parsing (I have had problems with DB
configuration strings in the file that work fine on the command line).
The good thing about this server is the support for ephemeral
credentials (if you create a web-service to generate them). This is a
necessity for WebRTC as the alternative is often to embed the TURN
credentials in the Javascript.
Regards,
Peter
Name: turnserver
Version: 1.8.3.6
Release: 0%{dist}
Summary: RFC5766 TURN Server
Group: System Environment/Libraries
License: BSD
URL: https://code.google.com/p/rfc5766-turn-server/
Source0: https://rfc5766-turn-server.googlecode.com/files/%{name}-%{version}.tar.gz
Patch00: turnserver-1.8.3.6-CentOS.patch
BuildRequires: gcc, make, redhat-rpm-config
BuildRequires: openssl-devel, libevent-devel = 2.0.0, mysql-devel
BuildRequires: postgresql-devel
Requires: openssl, libevent = 2.0.0, mysql-libs, postgresql-libs
%description
The TURN Server is a VoIP media traffic NAT traversal server and gateway. It
can be used as a general-purpose network traffic TURN server/gateway, too.
This implementation also includes some extra features. Supported RFCs:
TURN specs:
- RFC 5766 - base TURN specs
- RFC 6062 - TCP relaying TURN extension
- RFC 6156 - IPv6 extension for TURN
- Experimental DTLS support as client protocol.
STUN specs:
- RFC 5389 - base new STUN specs
- RFC 5769 - test vectors for STUN protocol testing
- RFC 5780 - NAT behavior discovery support
The implementation fully supports UDP, TCP, TLS and DTLS as protocols between
the TURN client and the TURN Server. Both UDP and TCP relaying are supported.
Flat files, MySQL or PostgreSQL are supported for the user repository (if
authentication is required). Both short-term and long-term credentials
mechanisms are supported.
For WebRTC applications, TURN Server REST API for time-limited secret-based
authentication is implemented.
The load balancing can be implemented either by external networking tools, or
by the built-in ALTERNATE-SERVER mechanism.
The implementation is supposed to be simple, easy to install and configure. The
project focuses on performance, scalability and simplicity. The aim is to
provide an enterprise-grade TURN solution.
To achieve high performance and scalability, the TURN server is implemented
with the following features:
- High-performance industrial-strength Network IO engine libevent2 is used
- Configurable multi-threading model implemented to allow full usage of
available CPU resources (if OS allows multi-threading)
- Multiple listening and relay addresses can be configured
- Efficient memory model used
- The TURN project code can be used in a custom proprietary networking
environment. In the TURN server code, an abstract networking API is used.
Only couple files in the project have to be re-written to plug-in the TURN
server into a proprietary environment. With this project, only implementation
for standard UNIX Networking/IO API is provided, but the user can implement
any other environment. The TURN server code was originally developed for a
high-performance proprietary corporate environment, then adopted for UNIX
Networking API
- The TURN server works as a user space process, without imposing any special
requirements on the system
%package utils
Summary: TURN Server utils and client development tools
Group: Development/Libraries
Requires: %{name} = %{version}-%{release}
%description utils
This package contains the TURN server utils and client development tools.
%package doc
Summary: TURN Server documentation and examples
Group: Development/Libraries
Requires: %{name} = %{version}-%{release}
BuildArch: noarch
%description doc
This package contains the TURN server documentation and examples.
%prep
%setup -q -n %{name}-%{version}
%patch00 -p1
%build
PREFIX=/usr CONFPREFIX=%{_sysconfdir} EXAMPLESDIR=%{_datadir}/%{name} \
MANPREFIX=%{_datadir} LIBDIR=%{_libdir} ./configure
make
%install
rm -rf $RPM_BUILD_ROOT
DESTDIR=$RPM_BUILD_ROOT make install
mkdir -p $RPM_BUILD_ROOT/%{_sysconfdir}/rc.d/init.d
install -m755 centos/turnserver.init \
$RPM_BUILD_ROOT/%{_sysconfdir}/rc.d/init.d/turnserver
mkdir -p $RPM_BUILD_ROOT/%{_sysconfdir}/sysconfig
install -m644 centos/turnserver.sysconfig \
$RPM_BUILD_ROOT/%{_sysconfdir}/sysconfig/turnserver
%clean
rm -rf $RPM_BUILD_ROOT
%pre
%{_sbindir}/groupadd -r turnserver 2 /dev/null || :
%{_sbindir}/useradd -r -g turnserver -s /bin/false -c TURN Server daemin -d \
%{_datadir}/%{name} turnserver 2
Re: [SR-Users] Kamailio for Debian blog?
On 13/05/13 12:24, James Cloos wrote: DP == Daniel Pocock dan...@pocock.com.au writes: DP I'd like to write a brief blog about the status of WebRTC in Debian, DP with a focus on SIP DP I understand Kamailio 4.0.1 is already in unstable, is that recommended DP for potential WebSocket users? The control file used for deb's packaging of 4.0.x does not include the tls, outbound or websocket modules. They provide a separate control.tls file one can use locally to compile and package kamailio with support for those modules. Ok, I see the procedure documented in README.Debian. I think this could be made much smoother for people by simply creating a TLS branch in the packaging SVN repository. Then people could just checkout the branch and run dpkg-buildpackage. The branch would be even easier to maintain if it is converted to git-buildpackage. For Fedora users, it can obviously be supported by conditional logic in the spec file, and then people can just run rpmbuild on the source tarball. There would need to be some flag that is passed on the rpmbuild command line to indicate whether the build is with or without TLS, it is probably OK to default build with TLS. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
Hi Daniel, 2013/5/13 Daniel Pocock dan...@pocock.com.au: Ok, I see the procedure documented in README.Debian. I think this could be made much smoother for people by simply creating a TLS branch in the packaging SVN repository. Then people could just checkout the branch and run dpkg-buildpackage. The branch would be even easier to maintain if it is converted to git-buildpackage. I'm going to migrate Debian kamailio repository to git ASAP. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] how to install kamailio with websocket function and msrp function on ubuntu？
Hi, this gist should work: https://gist.github.com/jesusprubio/4066845 2013/5/8 Daniel-Constantin Mierla mico...@gmail.com Hello, look on the wiki for tutorials of installing kamailio (those for debian should just work on ubuntu). Then read the readme files for msrp and websocket modules, they have sample config snippets inside. Cheers, Daniel On 5/7/13 3:50 AM, 李启明 wrote: hi, i am chinese,i am not good at kamailio.i want to install kamailio with websocket function and msrp function on ubuntu,could you give me a help？ -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] 4.0 forking behaviour
Hello, Has something changed about default forking behaviour in = 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
Hello, I don't recall any change in this aspect, are the two branches going to same destination? Cheers, Daniel On 5/13/13 2:43 PM, Alex Balashov wrote: Hello, Has something changed about default forking behaviour in = 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
Yes, they are identical in every way except for the .1 and .2 branch IDs. Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I don't recall any change in this aspect, are the two branches going to same destination? Cheers, Daniel On 5/13/13 2:43 PM, Alex Balashov wrote: Hello, Has something changed about default forking behaviour in = 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
Just to understand exactly: A calls B B redirects to C and is captured by proxy then from proxy you have two parallel outgoing branches to C? How you take the address of C and create the branch? uac_redirect or other script functions? Cheers, Daniel On 5/13/13 3:26 PM, Alex Balashov wrote: Yes, they are identical in every way except for the .1 and .2 branch IDs. Daniel-Constantin Mierla mico...@gmail.com wrote: Hello, I don't recall any change in this aspect, are the two branches going to same destination? Cheers, Daniel On 5/13/13 2:43 PM, Alex Balashov wrote: Hello, Has something changed about default forking behaviour in = 4.0? I have a scenario where INVITEs processed by the proxy first hit a redirect server, catch a 302, and then append another branch and iterate over one or more outbound routes. In the past, this worked fine. After I upgraded to 4.0, I am seeing two branches at a time on the outbound routes, after the initial branch to the redirect server. The desired behaviour is serial forking at all times. tm:failure_reply_mode is set to 3, as it always has been. Any ideas would be appreciated; thank you! -- Alex -- Sent from my mobile, and thus lacking in the refinement one might expect from a fully fledged keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] how to install kamailio with websocket function and msrp function on ubuntu？
Hi Jesus, thanks for sharing the config, will help many looking into this kind of features. I wonder if you had the time to test using the internal connections map instead of htable module, like in: - http://kamailio.org/docs/modules/stable/modules/msrp.html#idp120032 It will simplify the routing block for msrp a bit. Cheers, Daniel On 5/13/13 1:41 PM, Jesús Pérez Rubio wrote: Hi, this gist should work: https://gist.github.com/jesusprubio/4066845 2013/5/8 Daniel-Constantin Mierla mico...@gmail.com mailto:mico...@gmail.com Hello, look on the wiki for tutorials of installing kamailio (those for debian should just work on ubuntu). Then read the readme files for msrp and websocket modules, they have sample config snippets inside. Cheers, Daniel On 5/7/13 3:50 AM, 李启明 wrote: hi, i am chinese,i am not good at kamailio.i want to install kamailio with websocket function and msrp function on ubuntu,could you give me a help？ -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda http://twitter.com/#%21/miconda -http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 *http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com http://www.quobis.com/ -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
Hello Jesus, On 5/13/13 11:43 AM, Jesús Pérez Rubio wrote: Hi Daniel, we've developed a Javascript SIP stack which supports WebRTC have you published any out-of-the-box phone built from your stack? Something like one can take and in few config steps it can get the phone on their web page, without needing to code java script. Cheers, Daniel and we use Debian as base OS and Kamailio as SIP proxy. Some notes about the enviroment in case it could help you: - Kamailio stable (4.0) version included in official repo works fine. - Site: http://www.kamailio.org/wiki/packages/debs#latest_kamailio_40_release - Howto: https://quobis.atlassian.net/wiki/display/QoffeeSIP/Server+configurations - I've tested it with QoffeeSIP and JsSIP some days ago and there is no problem. - I've been playing with resiprocate-turn-server package but I had problems. It could be related with our client but we should take a look. I didn't give a try to Google TURN server but I'm going to do it, I'll keep you updated. PS: I'm co-mentor with you in GSoC, so we can speak about in in IRC channel. 2013/5/13 Daniel Pocock dan...@pocock.com.au mailto:dan...@pocock.com.au I'd like to write a brief blog about the status of WebRTC in Debian, with a focus on SIP I understand Kamailio 4.0.1 is already in unstable, is that recommended for potential WebSocket users? Has anybody else written any quickstart blog about WebRTC with that particular version, possibly with examples that are consistent with the Debian usage? For client side, SIPml5 is packaged, and I've had discussions with the JsSIP guys about packaging. For TURN, has anybody tried the TURN server project from Google code? It appears more advanced than the existing two TURN servers in Debian (e.g. it has database-backed authentication) http://code.google.com/p/rfc5766-turn-server/ and there is a package in progress: http://mentors.debian.net/package/rfc5766-turn-server ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com http://www.quobis.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
On 05/13/2013 09:30 AM, Daniel-Constantin Mierla wrote: Just to understand exactly: A calls B B redirects to C and is captured by proxy then from proxy you have two parallel outgoing branches to C? How you take the address of C and create the branch? uac_redirect or other script functions? Yes, your understanding of the scenario is correct. No, I do not use any of the uac_* or contacts functions. I manually catch the 302 in a failure_route, manually parse out the relevant details from the Contact header, rewrite the RURI (prior to append_branch()), append_branch() and t_relay(). -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
On 5/13/13 4:38 PM, Alex Balashov wrote: On 05/13/2013 09:30 AM, Daniel-Constantin Mierla wrote: Just to understand exactly: A calls B B redirects to C and is captured by proxy then from proxy you have two parallel outgoing branches to C? How you take the address of C and create the branch? uac_redirect or other script functions? Yes, your understanding of the scenario is correct. No, I do not use any of the uac_* or contacts functions. I manually catch the 302 in a failure_route, manually parse out the relevant details from the Contact header, rewrite the RURI (prior to append_branch()), append_branch() and t_relay(). append_branch() is not needed anymore (for couple of releases, actually, being added in one of the 3.x releases), but should be harmless unless you do other changes of r-uri/dst-uri after append_branch(). Can you try without append branch? Also, can you look at config execution trace to be sure append branch is not called twice somehow? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
On 05/13/2013 10:45 AM, Daniel-Constantin Mierla wrote: append_branch() is not needed anymore (for couple of releases, actually, being added in one of the 3.x releases), but should be harmless unless you do other changes of r-uri/dst-uri after append_branch(). Can you try without append branch? Oh. Well, that was news to me. I guess I missed this. So, now when t_relay() is called from failure route it automatically appends a new branch as necessary? Given this knowledge, I find it likely that branches are being automatically appended by the proxy and then additionally appended manually. I'll look into it from this angle. Thank you! -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
On 5/13/13 4:47 PM, Alex Balashov wrote: On 05/13/2013 10:45 AM, Daniel-Constantin Mierla wrote: append_branch() is not needed anymore (for couple of releases, actually, being added in one of the 3.x releases), but should be harmless unless you do other changes of r-uri/dst-uri after append_branch(). Can you try without append branch? Oh. Well, that was news to me. I guess I missed this. So, now when t_relay() is called from failure route it automatically appends a new branch as necessary? Yes, if the r-uri is changed in failure route, a new branch is created by t_relay() without need of using explicitly append_branch(). But again, using append_branch() once before t_relay() there should do nothing (backward compatible behavior). Unless some new code changed that behavior -- iirc, append_branch() was affected by some outbound code. Cheers, Daniel Given this knowledge, I find it likely that branches are being automatically appended by the proxy and then additionally appended manually. I'll look into it from this angle. Thank you! -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
Hi Daniel, We have something like you're asking for in the Github repository. First lines of this Quickstart guide show it ( https://quobis.atlassian.net/wiki/display/QoffeeSIP/Quick+start+guide). Only two steps are needed: - Clone the repo: *git clone https://github.com/Quobis/QoffeeSIP.git* - Copy examples/webphone/dist/* content to your Apache server. It's a simple webphone that we use to develop the stack. Another simplest one is also included in examples folder to help web developers to include it in their site. Nothing else, we're here if somebody needs something. Regards. :) 2013/5/13 Daniel-Constantin Mierla mico...@gmail.com Hello Jesus, On 5/13/13 11:43 AM, Jesús Pérez Rubio wrote: Hi Daniel, we've developed a Javascript SIP stack which supports WebRTC have you published any out-of-the-box phone built from your stack? Something like one can take and in few config steps it can get the phone on their web page, without needing to code java script. Cheers, Daniel and we use Debian as base OS and Kamailio as SIP proxy. Some notes about the enviroment in case it could help you: - Kamailio stable (4.0) version included in official repo works fine. - Site: http://www.kamailio.org/wiki/packages/debs#latest_kamailio_40_release - Howto: https://quobis.atlassian.net/wiki/display/QoffeeSIP/Server+configurations - I've tested it with QoffeeSIP and JsSIP some days ago and there is no problem. - I've been playing with resiprocate-turn-server package but I had problems. It could be related with our client but we should take a look. I didn't give a try to Google TURN server but I'm going to do it, I'll keep you updated. PS: I'm co-mentor with you in GSoC, so we can speak about in in IRC channel. 2013/5/13 Daniel Pocock dan...@pocock.com.au I'd like to write a brief blog about the status of WebRTC in Debian, with a focus on SIP I understand Kamailio 4.0.1 is already in unstable, is that recommended for potential WebSocket users? Has anybody else written any quickstart blog about WebRTC with that particular version, possibly with examples that are consistent with the Debian usage? For client side, SIPml5 is packaged, and I've had discussions with the JsSIP guys about packaging. For TURN, has anybody tried the TURN server project from Google code? It appears more advanced than the existing two TURN servers in Debian (e.g. it has database-backed authentication) http://code.google.com/p/rfc5766-turn-server/ and there is a package in progress: http://mentors.debian.net/package/rfc5766-turn-server ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-us...@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Jesús Pérez VoIP Engineer at Quobis Fixed: +34 902 999 465 Site: http://www.quobis.com ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Does Kamailio support PRACK ?
Hi, Does kamailio support PRACK method ? Any configuration change is needed?It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow: A send INVITE -- Kamailio -- proxy the packet to BB send 180 Ringing -- Kamailio - proxy the packet to AA send PRAK (increase Cseq) -- Kamailio -- proxy the packet to BB send 200 OK -- Kamailio - proxy the packet to AKamailio re-send 180 Ringing to A Thanks,AS ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Does Kamailio support PRACK ?
PRACK is an in-dialog/sequential request, which will be relayed like other. So, Kamailio relays it in the same way that it supports any sequential request. Is it not necessary for a SIP proxy to do anything specific to support PRACK. The UAC should not be increasing the CSeq when sending a PRACK. My guess is that it is the UAS which doesn't like it, rather than Kamailiio. From RFC 3262 Section 3 (UAS Behavior): A matching PRACK is defined as one within the same dialog as the response, and whose method, CSeq-num, and response-num in the RAck header field match, respectively, the method from the CSeq, the sequence number from the CSeq, and the sequence number from the RSeq of the reliable provisional response. -- Alex On 05/13/2013 11:37 AM, Alex Solt wrote: Hi, Does kamailio support PRACK method ? Any configuration change is needed? It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow: A send INVITE -- Kamailio -- proxy the packet to B B send 180 Ringing -- Kamailio - proxy the packet to A A send PRAK (increase Cseq) -- Kamailio -- proxy the packet to B B send 200 OK -- Kamailio - proxy the packet to A Kamailio re-send 180 Ringing to A Thanks, AS ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Does Kamailio support PRACK ?
Alex Solt writes: Does kamailio support PRACK method ? Any configuration change is needed? my understanding is that prack is like any on-dialog request and does not need any special handling on kamailio. i have not had any problems with pracks and there is nothing prack specific in my config. this is without dialog module which i don't know anything about. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Does Kamailio support PRACK ?
Just to confirm:Kamaili is forwarding the PRAK packet. However, it seems the kamailio ignore the Cseq increase within the PRAK and therefore it ignore the 200 Ok after that. Then, Kamailio re-send 180 again. Thank,AS Date: Mon, 13 May 2013 11:41:09 -0400 From: abalas...@evaristesys.com To: sr-users@lists.sip-router.org Subject: Re: [SR-Users] Does Kamailio support PRACK ? PRACK is an in-dialog/sequential request, which will be relayed like other. So, Kamailio relays it in the same way that it supports any sequential request. Is it not necessary for a SIP proxy to do anything specific to support PRACK. The UAC should not be increasing the CSeq when sending a PRACK. My guess is that it is the UAS which doesn't like it, rather than Kamailiio. From RFC 3262 Section 3 (UAS Behavior): A matching PRACK is defined as one within the same dialog as the response, and whose method, CSeq-num, and response-num in the RAck header field match, respectively, the method from the CSeq, the sequence number from the CSeq, and the sequence number from the RSeq of the reliable provisional response. -- Alex On 05/13/2013 11:37 AM, Alex Solt wrote: Hi, Does kamailio support PRACK method ? Any configuration change is needed? It appears Kamailio does not like the PRAK when increasing Cseq. Here is the call flow: A send INVITE -- Kamailio -- proxy the packet to B B send 180 Ringing -- Kamailio - proxy the packet to A A send PRAK (increase Cseq) -- Kamailio -- proxy the packet to B B send 200 OK -- Kamailio - proxy the packet to A Kamailio re-send 180 Ringing to A Thanks, AS ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Does Kamailio support PRACK ?
On 05/13/2013 11:49 AM, Alex Solt wrote: Just to confirm: Kamaili is forwarding the PRAK packet. However, it seems the kamailio ignore the Cseq increase within the PRAK and therefore it ignore the 200 Ok after that. Then, Kamailio re-send 180 again. Incorrect on all counts. 1) Kamailio does not ignore anything in the PRACK; 2) Kamailio cannot ignore 200 OKs; it simply passes all replies that it receives. It is not a recipient of 200 OKs. Its job is to pass them back to the user agent that is. 3) Kamailio likewise cannot originate replies, so it does not re-send the 180. The UAS does. In other words, the problem you are seeing is between the endpoints that are calling each other through the proxy. Kamailio is just a dumb, disinterested messenger here. -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] kamailio 4.0.1 and nokia e72 udp
It doesn't seem to be the router/NAT's problem though, as the Nokia itself binds to the right port at first, then gives up on it and changes to a port 20 higher instead. The second bind is also the one that it advertises in it's sdp. But that tip with listen for port changes is good, it would only be problematic if there are multiple concurrent calls from the same (perhaps NATted) IP, right? On 5/13/13, Andres and...@telesip.net wrote: On 5/11/2013 4:29 PM, hiro wrote: using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind NAT registered via UDP I get no voice. The e72 strangely sends a single udp packet from a wrong port (49152) before the rtp stream should start. This quirk of the e72 doesn't seem to work well with rtpproxy if the following analysis is true: rtpproxy detects that single UDP packet from the wrong port and so we think that is where everything else will also come from and stop listening on other ports. we then also answer on that wrong port. Although all subsequent incoming packets arrive from the expected (49172) port sent also in the sdp and to the right one we had sent in the sdp earlier we never receive them, because we still listen on that wrong port with that one bogus packet. I have seen such behavior before from other cheap NAT routers. The solution was to keep rtpproxy in listen mode for port changes always. That way it will keep working no matter how many times the port changes on the client side. We are still running an older version of rtpproxy so I cannot comment on how to patch the lastest version. The version we have is 1.0.2 and the modification we did was to file main.c and commented the following aroubd line 1415: /*sp-canupdate[ridx] = 0;*/ Thats it. -- Technical Support http://www.cellroute.net ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
On 13/05/13 16:58, Jesús Pérez Rubio wrote: Hi Daniel, We have something like you're asking for in the Github repository. First lines of this Quickstart guide show it ( https://quobis.atlassian.net/wiki/display/QoffeeSIP/Quick+start+guide). Only two steps are needed: - Clone the repo: *git clone https://github.com/Quobis/QoffeeSIP.git* - Copy examples/webphone/dist/* content to your Apache server. It's a simple webphone that we use to develop the stack. Another simplest one is also included in examples folder to help web developers to include it in their site. Ok, so my blog went up earlier today, thanks for all the feedback, I've linked to this thread too: http://danielpocock.com/get-webrtc-going-fast The main aim was to show the quickest way to get started - so I introduce it with the repro packages but Kamailio is covered too. When repro graduates from experimental and Kamailio packages have a streamlined TLS install, I'll do another blog to hopefully alert more people to test it all. Jesús, have you tested QoffeeSIP with repro yet? Feel free to hassle us on the repro or reSIProcate lists if it doesn't work. ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
On 5/13/13 8:35 PM, Daniel Pocock wrote: On 13/05/13 16:58, Jesús Pérez Rubio wrote: Hi Daniel, We have something like you're asking for in the Github repository. First lines of this Quickstart guide show it ( https://quobis.atlassian.net/wiki/display/QoffeeSIP/Quick+start+guide). Only two steps are needed: - Clone the repo: *git clone https://github.com/Quobis/QoffeeSIP.git* - Copy examples/webphone/dist/* content to your Apache server. It's a simple webphone that we use to develop the stack. Another simplest one is also included in examples folder to help web developers to include it in their site. Ok, so my blog went up earlier today, thanks for all the feedback, I've linked to this thread too: http://danielpocock.com/get-webrtc-going-fast The main aim was to show the quickest way to get started - so I introduce it with the repro packages but Kamailio is covered too. An alternative for installing kamailio with tls is to use kamailio.org repositories for debian distros. Might be easier for many than recompiling. Thanks for spreading the word around the world, Daniel When repro graduates from experimental and Kamailio packages have a streamlined TLS install, I'll do another blog to hopefully alert more people to test it all. Jesús, have you tested QoffeeSIP with repro yet? Feel free to hassle us on the repro or reSIProcate lists if it doesn't work. -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013 * http://asipto.com/u/katu * ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio for Debian blog?
DP == Daniel Pocock dan...@pocock.com.au writes: DP I'd like to write a brief blog about the status of WebRTC in Debian, DP with a focus on SIP DP I understand Kamailio 4.0.1 is already in unstable, is that recommended DP for potential WebSocket users? The control file used for deb's packaging of 4.0.x does not include the tls, outbound or websocket modules. They provide a separate control.tls file one can use locally to compile and package kamailio with support for those modules. The issue is openssl. Evidently kamailio does not support gnutls? Debian is unable to distribute binaries of kamailio linked to openssl because openssl's license is not gpl-compatible and kamailio does not have a linking exception which would permit distribution of such binaries. To get kamailio's websocket support into debian proper, kamailio needs either to work with a gpl-compatible tls library (openssl may be the only one which is not) or it needs to add a linking exception to its license to permit binary distribution when linked with openssl. There is a note at: http://www.gnome.org/~markmc/openssl-and-the-gpl.html discussing the issue. The wikipedia page on openssl mentions wget as an example of a gpl'ed package with such a linking exception. This is likely to be an issue for other binary dists, such as fedora. -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 4.0 forking behaviour
Daniel, Thank you for your help. FYI, stopping my use of append_branch() everywhere solved the problem. I was unaware that it had become an essentially deprecated requirement. Thanks again! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] event_route
Hi All, in a call forking, after one branch answer the call (200 OK reply), a CANCEL SIP message has been sending to other/another branch(es) and I need to process this/these cancellations in configuration file. After reading some documentations, I discovered there is event_route[tm:local-request] block, which is executed when tm generates internally and sends a SIP request, Such cases are: SIP messages sent by msilo module SIP messages sent by presence server SIP messages sent by dialog module SIP messages sent via MI or CTL interfaces I didn't understand very well this cases, so I insert event_route block in my kamailio.cfg but neither CANCEL SIP message or other requests generated by tm module was handled by event_route. I must be using wrong concept to handle this CANCEL SIP message, it's possible handle this messages in configuration file? Best Regards ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
