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
Hi Daniel, we've developed a Javascript SIP stack which supports WebRTC 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 daniel@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
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 <daniel@pocock.com.au mailto:daniel@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
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 miconda@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 daniel@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-users@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
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
Hi Jesus,
one more newbie qoffeesip question :-) -- does this have to be installed separately (I mean qoffeesip)? Or is part of your repository as well?
Thanks, Daniel
On 5/13/13 4:58 PM, 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.
Nothing else, we're here if somebody needs something.
Regards. :)
2013/5/13 Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@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 WebRTChave 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, Danieland 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 <daniel@pocock.com.au <mailto:daniel@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 <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://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/
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.
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.
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
"DP" == Daniel Pocock daniel@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
On 13/05/13 12:24, James Cloos wrote:
"DP" == Daniel Pocock daniel@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.
Hi Daniel,
2013/5/13 Daniel Pocock daniel@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.
2013/5/13 Victor Seva linuxmaniac@torreviejawireless.org:
I'm going to migrate Debian kamailio repository to git ASAP.
Well, finally I've created the git repository with git-buildpackage importing just the 3.0.1-1 and 4.0.1-1 versions.
I think it's enough
-
Daniel Pocock -
Daniel-Constantin Mierla -
James Cloos -
Jesús Pérez Rubio -
Peter Dunkley -
Victor Seva