Documentation Index
FAQ
Linux LiveCD VoIP Router http://www.wifi.com.ar/english/voip/
- Why use our solution ?
It is a proven bundle of software with over 50,000 active
users and 200+ server installations.
It uses SER for scalable sip phone registration, control
and user web interface, and optionally Asterisk for B2BUA.
It has email technical support and updates.
You can be up and running in days.
- What do I need to get started ?
You need a PC with a fixed public ip connected to the internet.
Also your internet router needs a fixed ip (usually business
internet connections provide this). Or you can host it at a
datacenter, it requires a dedicated server that can boot from
a livecd. In addition you need some basic linux kwnoledge.
- Why not ip with port forwarding ?
You need a public fixed ip on your voip server, no port forwarding.
In general business internet connections can provide this, also
datacenters. A fixed ip on your internet router plus and additional
fixed ips for internal servers.
Port forwarding can not be used, nor firewalling between your
internet connection and the voip server. Either of these will
break nat traversal and more services on the voip server.
For these reasons in general consumer adsl internet connections
can not host the voip server.
- Does it support Hard Disk Install ?
Yes. You can install to hard disk.
- Does it support VMware
Yes. The software can be run from VMware
- LiveCD Reliability
We have over 250 server installations running. Some servers have
more than 30,000 active users and have been up for 12 months
continuously.
- How much bandwidth per call do I need ?
None on the default configuration. Which only does
call setup and acounting. The sip phone connects
directly to the voip gateway.
If you use the asterisk b2bua (included as an extra module),
then you need to carry the traffic of the rtp streams on your
server. Or on the geographically distributed RTP proxy servers.
- What computer do I need ?
We are running fonosip.com with an AMD Duron 2000 and 512 MBytes
of ram. With 10,000 active users and a cpu average usage under
10%. So the answer is: you do not need a big machine
- What hard disk do I need ?
You can use ide or sata.
No support for disk arrays or raid.
We recommend ide. you can do network replication of your database
data for security
- What does it mean no B2BUA in base package ?
The base solution does not include back to back user agent
(B2BUA). Thus you can not cut a call in progress.
But it is not a problem for monthly or business users, or credit
card users (Fonosip.com style business).
In addition it allows the sip phone to connect
directly to the pstn gateway, after autentication and accounting.
Lowering your bandwidth bill and lowering delay and latency for
the user.
For providing wholesale or call shop services, or suing multiple
external pstn termination providers, it is recommended
to use the extra module asterisk b2bua.
- What computer do I need for B2BUA ?
An AMD 2000 mHz with 1 GByte of RAM will do 200 simultaneous
calls without call degradation. Without doing any codec
transacoding.
- Which codecs are included ?
The base solution is a sip proxy, so the codecs are controlled
by the sip phone and the pstn gateway.
In the B2BUA astersik solution you have G711, gsm, ilbc. And can
do pass-through g723, g729
In addition you can buy G729 liceses from Digium at $10 per channel
- What version of linux do I need ?
None, linux is included in the livecd. You don't need to
install anything.
- What size hard disk do I need ?
5 Gbytes or larger.
- Can I run it without a hard disk ?
Yes, but mainly for demo and testing purposes, since you will
not be able to keep call accounting or user database in case of
a power loss.
An RTP proxy box may run without a had disk, saving config to
floppy.
- Can I customize the web interface ?
Yes. All html and php sources are included and can be modified
- Can I do SER high availability ?
Yes, using a second server machine and
DNS SVR Resource Records. But note that not all UAs support
this.
Also the livecd solution comes in handy when managing several
boxes of SER/Asterisk, since you can build/rebuild servers in
minutes.
- Can I do RTP Proxy high availability ?
Yes, using multiple machines for RTP Proxy. Can also be
geographically distributed.
- What happens if I loose a BYE message ?
If you loose a bye message from your gateway, or a sip phone
crashes, and you do not use B2BUA:
you need to make sure your gateway has an rtp timeout.
If you use a cisco gateway such cisco 5300 make sure you run
the latest IOS, and set rtptimeout to 1 minute or so.
That way you will not miss any calls on your accounting.
Or get any runaway calls.
Good voip termination providers in this respect are
fonosip.com, voip.brujula.net (of course :)
But under some UA / Gateway combinations you may always loose
some BYEs (maybe 1%). To avoid that completely you need a B2BUA
(provided in the extra module asterisk b2bua)
- What PSTN gateways can I use ?
Any SIP compliant gateway will do. Cisco as5300, as5350, as5800,
quintum, asterisk, in general any SIP compatible gateway.
- When do I need the Asterisk B2BUA module ?
If you will use your own cisco 5300s, or you control all of
your pstn gateways, then the base system might be enough for
you.
The additional b2bua is required when you use many different
gateways, and some do not support rtp timeout. In this case
to guarantee your CDRs you need the B2BUA.
You also need this module if you want to do RTP geographycal
distribution (to guarantee QoS), or codec transcoding.
- Is there any way to try/test/demo the functionalities of
this solution for some limited time?
There is a live demo site at FonoSIP.com.
But we do not have a demo site for the admin side of the
solution. Just the screenshots on the web.
There is no limited time demo product.
- How is the software delivered ?
It is a download
- What is your refund policy ?
We have a money back guarantee for 30 days.
And a 10% refund fee.
- What domain name should I use
If you want your users to be
81234@mydomain.com
you must point your dns
mydomain.com ==> ip.of.your.server
if you can not do that, then use something like
voip.mydomain.com ==> ip.of.your.server
- Do you have a distributor program ?
No, unfortunately we only sell direct
- Benefits of Linux VoIP Server compared to Asterisk ?
NAT Traversal
Linux VoIP Server deals a lot better with NAT traversal.
SER allows you to send the voice (or video stream) from
your IP Telephone or VoIP software client directly to a
VoIP telephone service provider using in most cases
(non-symmetric NAT). SER also allows you to directly
manipulate SIP communication to handle special cases,
such as, when you have two VoIP telephones behind the
same NAT router and want to send the media directly
between them.
Load Balancing
A unique strength of SER is its ability to load balance
VoIP calls. Specialized hashing algorithms in SER can be
configured to load balance by the "username", "ruri",
"callid", and other properties. SER is "failover aware"
and can make for a very complementary part of an
Asterisk solution.
PSTN Cards:
SER always need a SIP gateway to connect to the PSTN.
There is no possibility to install telephony cards in the
server. Often, we will implement Asterisk as a PSTN
gateway for SER.
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