Portable VOIP network?

I am taking over a rather strange project, and thats why I have some strange demands for our new voip solution.

My team and I will for the next 18 months do several trips in europe and usa. Still we will need to have a callcenter active during business hours. Our callcenter team will travel along with the rest, and for the most part contain 10 people but should scale upward to max 30 people.

We have gotten tips to use freepbx and
Crystal Racording Interface for our recording needs.

What we need is system that we can bring along from one place to the next. We belive we will run softwarephones from laptops rather then handheld phones.

We need the system to handle recording of all calls. We also need to find calls so system must show dialed number, date and time for each call and so on.

From what I can read the hardware needed is not too extreme. So we plan to use small formfactor cases with Core 2 duo laptop cpu to run it all from. Here we will use as meny as needed. We will need a switch with 30+ ports and connect the laptops to the network. Am I crazy or is this basically what we need?

Any ideas to how hard something like this would be to set up?

To sum it all up:

1: What kind of hardware are we looking at here?

2: Possible to record all calls made with number called, time and date?

3: Possible to do this with softwarephones on laptops? pls give me links to desent swphones.

4: We have a internal crm-like system that handles all our customer base could this system be integrated with the VOIP system?

5: If no 4 is possible the callcenter could stick to one system and have sort of a call-button inside the application that launches the VOIP and the operators uses headset to make calls directly from laptop.

6: What kind of bandwidth consumption are we looking at for a system like this?

7: Finally is there anything that will be a problem with an approach like this?

I hope to get lots of valuable feedback here and would like to here from others that might have done something similar.

thx- now of to read up on voip

willie

Greetings xpwill and welcome to VoIPUser forum.

xpwill :

Still we will need to have a callcenter active during business hours. Our callcenter team will travel along with the rest, and for the most part contain 10 people but should scale upward to max 30 people.

Are you expecting your call-center system to service a concurrent 30 calls? If not, you may want to scale it down using some embedded system.

Currently, I have my asterisk system running on my (underpowered) LaFonera WiFi router. It isn't a sophisticated system as you had described on your post above; however, it is a portable system that can use any freely accessible WiFi hotspots. It has been configured with more than 50 VoIP channels from various free (non-commercial) VoSPs with several extensions. So far, several people and I (about three) are the only ones who use the system. So, the load to serve 50 channels is pretty minimal. The system has no problems to serve five concurrent calls with no transcoding involved.

The process of logging callerID's and date time stamps is known as CDR (Call Detail Records) logging. Most PBX type systems have something which can do this. Asterisk is an option that you haven't mentioned, and there is a free billing engine available called a2billing which may be worth looking at (for CDR collation).

For softphones, take a look at www.counterpath.com and www.zoiper.com

Most of the conference call systems that run on Asterisk have some means of recording those, so might be some (free/opensource) code that you can utilise or have re-worked to solve that problem.

Dean

mazilo :

Greetings xpwill and welcome to VoIPUser forum. Are you expecting your call-center system to service a concurrent 30 calls? If not, you may want to scale it down using some embedded system. Currently, I have my asterisk system running on my (underpowered) LaFonera WiFi router. It isn't a sophisticated system as you had described on your post above; however, it is a portable system that can use any freely accessible WiFi hotspots. It has been configured with more than 50 VoIP channels from various free (non-commercial) VoSPs with several extensions. So far, several people and I (about three) are the only ones who use the system. So, the load to serve 50 channels is pretty minimal. The system has no problems to serve five concurrent calls with no transcoding involved.

We need to be able to run 10-12 concurrent calls yes. Sadly not much we can do with the number of operators in the callcenter. So that means we need to find the best possible solution. Also we might need the option to double the capcity
to run upwards to maybe 20 concurrent calls.

-thx

xpwill :

We need to be able to run 10-12 concurrent calls yes.

I am sure some embedded system will handle this easily. There is also some x86 micro based system that will be able to run an asterisk system with lots of whistles.

Thx Dean.

Yes CDR is importent to us. See allready picking up the lingo

Hm maybe the whole point of pbx's like asterisk and freepbx is to run a regular phonesystem from the server running the software? Is it at all possible to route voip traffic from the server via a switch directly to individual laptops?
Reasson for asking is that I spoke with a guy during lunch and when I mentioned my plan i was told it was not at all doable because the sw-pbx's is meant to have phonelines running from the server not lan-cables.

I do not know if this is at all the truth, or if there is ways to do what i scetched here before.

-thx

mazilo :

I am sure some embedded system will handle this easily. There is also some x86 micro based system that will be able to run an asterisk system with lots of whistles.

Thx mazilo.
Ok if you really think this is the route to go I will start reading up. do o have any links to articles or webistes that shows embedded systems?

I belive I read on asterisk site that converting the call in real-time from one standard into antoher codec or fileformat is puttting strain on the cpu, and this strain is almost linear with the number of concurrent users, do you know if this is correct?

Hi

firstly the guy that said

Quote:

i was told it was not at all doable because the sw-pbx's is meant to have phonelines running from the server not lan-cables.

may b=not be the best guy to to for advice to.

Basicly Asterisk running on a small "server" will handle all you want and linked to softphones will happily handle your load.

Something like a shuttle or the like, or even a decent spec laptop would do what you want. IBM thinkpad springs to mind.

Install PBXinaflash or the like and off you go,

Ian

mazilo :

I am sure some embedded system will handle this easily. There is also some x86 micro based system that will be able to run an asterisk system with lots of whistles.

ok have done some reading and what seems to be the case is that most of the processing or coding/decoding will actually take place on each individual laptop rather then on the server, is that correct? So a low end server or even embedded will easily serve 10-15 concurent calls?

I seen that the G.729 Codec is the codec that runs the most compressed (?) calls. What do you think the bandwidth consumption will be with 10 concurrent calls? (I belive i saw a bandwidth-calculator somewhere but can't find it back)

Finally any specific tips when it comes to embedded systems or micro based systems? Do remember that I will need to store each and every calls for about 6 months so I also need some storage space.

ianplain :

may b=not be the best guy to to for advice to. Install PBXinaflash or the like and off you go, Ian

You are abolutly correct there ianplain
Thx for introducing me to PBXinaflash.

will

Hi

Unless you can get the callls working in passthrough mode and the endpoints support g729 the work will be done on ther server.

Why not look at the digium appliance or the xorcom system or one of the other appliances. These will give you a guaranteed level of working.

How portable do you need it ? personaly I would go for a laptop as it will do all you want be low power and have the screen etc.

Ian

ianplain :

Hi Unless you can get the callls working in passthrough mode and the endpoints support g729 the work will be done on ther server. Why not look at the digium appliance or the xorcom system or one of the other appliances. These will give you a guaranteed level of working. How portable do you need it ? personaly I would go for a laptop as it will do all you want be low power and have the screen etc. Ian

Hmm, ok so the cpu-consumtion due to codeing/de-coding will be on the server if not custom-fitted or hammered into shape. Well here is a cost/benefit question. What will cost less take less time. Maybe better to spring for better grade hw then doing to much modification to the sw.

Well as we wil be moving from city to city during this project we need some portability. Hardcore watercooling for cpu and monster heatproducing cpu's are not an option. But as you said yourself a shuttle or other sff box will be perfect for us. For safety we could take it as carry-on on planes that way we will not loose critical equipment. Also I belive cooling will be easier with a sff pc then cooling a regular laptop. Regarding screen admin can remotecontrol the server from his laptop.

Is there any ways to put the core2duo's in the laptops to good use? How much of the processing could we actually make take place locally on laptops rather then on the server? If anyone has taken a similar route for a project pls let me know how you solved this.

will

Ok

You will be travelling, so you need standard kit.

So why not plump for a bunch of identical laptops one to be used as the voip server the rest for users. This way if the server hardware fails you can just swap the harddrive. take a ghost image of the "voip server" on a CD/DVD/spare harddrive so if you loose the Voip server you can use any of the others by ghosting back/swapping harddrive.

This will give you ultimate reliability and distaster recovery.
any modern laptop will handle 10 concurrent calls.

Ian

Found back to the bandwidth calculator

But the field of voip is somewhat hard to understand. I see there are different choiches for protcols. I belived the standard was SIP and I am not clear on what differs one from the other.

* MGCP
* H323
* SIP
* IAX2
* IAX2 trunked

Which one is the best to use and why?

will

Hi

Softphone use SIP its the most common.

For trunks for in/outgoing use either SIP/IAX
IAX will handle firewalls better.

TBH I would just go for SIP

Ian

xpwill :

bandwidth calculator ...

That's a useful tool.

Re: protocols to use. SIP/RTP is by far the most common combination due to large numbers of implementations and a lot of interop work. H.323/RTP is largely now confined to service providers. A lot of service providers are installing equipment to interface/migrate legacy H.323 to SIP, so in the long term H.323 will be nil. IAX is a proprietary protocol that comes along with the Asterisk PBX, so it's popular in that community and starting to spread to other applications on the periphery. As Ian says, it tunnels firewalls well, on account of signalling and media travelling together down the same "pipe".

MGCP is not really a contender. This is a protocol that allows a softswitch to control a media gateway, so is mostly used by service providers.