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Free SIP server (like ekiga, sip2sip) with websocket support, any interest?
As subject.
It will provide SIP service like ekiga.net, sip2sip.info,iptel.org.
Main difference is that it will support websocket. So you can use your browser to login.
In first phase, you can only register and make a call to another SIP endpoint.
It will not support presence, chat, and another advanced features.
I will add it when there are many requests to add it.
Disclaimer : i'm relatively new to SIP (+- 7 months) but i have strong interest in it.
So, there is a chance that the server still unstable in the beginning. I promise to do my best to provide good server.
Comments
Surely Interested, if it performance well. Right now, I ran my own Asterisk
nice i am intrested
You should probably describe what you want to do as free VoIP.
Free SIP implies free SIP Proxy and nothing else. I'm assuming that's only part of what you want to do.
I am also interested, currently running my own asterisk sip pbx on a 128mb ram open vz vps. My main use is 1) to register multiple incoming DID and 2) least cost routing based on the number dialled.
please provide more details of exactly what you are going to be doing so that I may help/offer suggestions.
So, running Asterisk on 128mb, does it really work? I guess you did not have freepbx installed. Asterisk itself does not consume much memory while idling. But those Apache/Mysql for Freepbx do. And if you have many active calls, it does not hold well.
yes it works happily in 128mb, it would even work on 96mb and probably 64mb with some burst/swap if I set up a cron job to reboot every 24-48 hours! It usually sits at around 60-90mb ram used but I have apache and a few other services running which I could get rid of and bring the ram usage down further. I am just using asterisk and a non-standard GUI called FIVN, if I use FreePBX I idle around 160-200mb!!! But I really have no use for freepbx so don't use it.. At most I have 2 active calls, but I am sure it could cope with many more, maybe ~30-40, I don't think the calls add that much to the ram used.
@dnwk are you running asterisk also? I have put up a script in the tutorials section if you want to have a go with it on centos 5.x 32 bit.
Yes, i have described it a bit in my first post.
It is not free VOIP, it is SIP server. Like ekiga.net and sip2sip.info.
In short, it will provide service for SIP to SIP call. There is no plan to support calling PSTN.
Sure it does:
asterisk 17506 0.0 5.5 119164 6784 ? Ssl Jul11 0:12 /usr/sbin/asterisk -p -U asterisk
You still don't seem to understand the difference between just SIP and VoIP. I don't think you are ready to offer a service if you don't understand the basics first. VoIP has nothing to do with whether you are calling a PSTN. ekiga.net describe themselves as a free VoIP service.
And if you allow me to put in an external trunk?
Great. Where do you put your FIVN? You run it on your own computer? Where can I get FIVN.
I run asterisk 11 on Ubuntu. I know most of them in the world run it on centos. Now, I pretty much try to avoid CentOS and keep everything to Ubuntu.
Where is the tutorial? I did not find it.
@dnwk it's in the tutorial section of this site, script only works on centos 5.x 32 bit http://www.lowendtalk.com/categories/tutorial
@siaku you need to clarify what you're going to be doing and what purpose is. If a similar service already exists which is free to use, then I'm not sure what the point of your service is. Also one of the most important things is being able to have sip trunks etc. It sounds like what you're doing something like a basic skype voip-voip service without any of the bells and whistles skype has.
Relatively easy to cobble together an Asterisk + XMPP skype/chat type service these days.
Well, 1.4.4 is really an old version of Asterisk. They have lots of problems.
1.4.44 is a LTS version and it works perfectly well.
Hi,
Thanks for your comment.
Yes, you're right.
In first phase it will only a basic SIP server where you can register into it and call another SIP phone.
I will also consider to provide another service like described here :
https://www.ekiga.net/index.php?page=services if there are needs for it, of course with limitation .
And maybe another service, depend on the request.
But i have no plan to provide service that require some billing. Because it will be only hobby project.
The main difference is that my server will support websocket. So you can use your web based sip phone to register, for example :
For sip trunk, i'm still not sure what it is. I will read about it.
Some of my providers have compatibility issue with 1.4. It might be their problems but after upgrade there is no issue. And I think Digium did not introduce the idea of LTS until 1.8
SIP trunk, it is, an external provider which you can direct your call into PSTN. So you won't need to do the billing. User find their own provider and its their responsibility to do the billing.
Do you have any experience configuring Callcentric? Their system configure is so difficult to configure
Thanks, it seems interesting.
I will try my best to provide it.
Once again, i'm still new, but i have very strong interest in it.
I even already sent a patch to some SIP lib and accepted, but it only 2 lines of trivial patch :P. The point is, i'm ready to learn, not only it's administration, but also the protocol and the implementation.
Thanks again.
No, both 1.4 and 1.8 are LTS versions, LTS for 1.4 ended in 2012, but many people are still using it because it is so stable. I have used 1.8 also, but like 1.4. Digium GUI works well with 1.4, support was added for 1.6 and 1.8, but some people report problems with digium gui and asterisk 1.8, that's why I stick with 1.4
@siaku - if we can register out our sip trunks (providers) you don't have to do billing and we can have incoming sip calls too.
If you are doing a new service you should be using Asterisk v1.11. Only reason to be using an older version is if you have some legacy issues you are dealing with.
v1.11 has features that older versions don't such as new version of confbridge conferencing app. Also fixed xmpp which is something anyone wanting to start a VoIP + chat service would probably want.
Agree. I was just posting about my love for 1.4.44, but the most recent asterisk should be used for a new project. 1.4.44 just works for me, and if I upgrade I'd have to mess around with the *.conf files which I can't be bothered with.
I love freeeswitch
Freeswitch can be good depending on what you are doing. Much bigger community with Asterisk which is an important considering when dealing with opensource.
You are going to need better linux skills with Freeswitch. There isn't as much information out there about it. However, the modular design makes it easy to tailor custom solutions. For a SIP to SIP VoIP service it has potential when combined with SIP Proxy or Kamilio. Asterisk is better at processing the calls which is more the traditional PBX functions. Things like transcoding, voicemail, IVR, Call Queues.