Is my skype based call center to become a monstrosity, or shall it carry us to the promised land?
May 8, 2008 6:27 PM Subscribe
Can I set up a temporary call-center with ten different lines using Skype? Or is there another VOIP solution that would cost less than installing new lines?
My office needs to make a lot of phone calls over the course of three weeks and then the number of calls will drop back down dramatically. Installing new physical lines is very expensive, especially since we won't be using them for long. So, I looked at the unlimited call plan that Skype has, and was giving it some very strong thought.
The basic plan so far is to set up 2 DSL lines running at 7 Meg, wire up a router and switch to each, run cable to ten computers, and hook up USB handsets to each one. The computers would be dedicated only to Skype usage for that time and the USB handsets would be used to dial out. Each computer would have a separate Skype account.
I hope that this would be enough bandwidth and that multiple Skype connections would play nicely with each other. But, I've only used Skype at home running one connection, and so I really don't know how it would work.
I'm open to other solutions, but the system needs to be ready to roll in about 7-10 days, and I have no experience setting up an Asterisk system or anything else. I once had a Vonage account and using that would be fine except that they require a contract.
I appreciate any advice you brilliant folks might give. Thanks!
My office needs to make a lot of phone calls over the course of three weeks and then the number of calls will drop back down dramatically. Installing new physical lines is very expensive, especially since we won't be using them for long. So, I looked at the unlimited call plan that Skype has, and was giving it some very strong thought.
The basic plan so far is to set up 2 DSL lines running at 7 Meg, wire up a router and switch to each, run cable to ten computers, and hook up USB handsets to each one. The computers would be dedicated only to Skype usage for that time and the USB handsets would be used to dial out. Each computer would have a separate Skype account.
I hope that this would be enough bandwidth and that multiple Skype connections would play nicely with each other. But, I've only used Skype at home running one connection, and so I really don't know how it would work.
I'm open to other solutions, but the system needs to be ready to roll in about 7-10 days, and I have no experience setting up an Asterisk system or anything else. I once had a Vonage account and using that would be fine except that they require a contract.
I appreciate any advice you brilliant folks might give. Thanks!
Response by poster: We just need to be making 10 calls concurrently, with no need for incoming calls. I should have clarified that the call center will only be making outgoing calls.
posted by abkadefgee at 6:49 PM on May 8, 2008
posted by abkadefgee at 6:49 PM on May 8, 2008
What's the upload speed on those DSL lines? That's the limiting factor on VOIP in this manner.
posted by sharkfu at 7:33 PM on May 8, 2008
posted by sharkfu at 7:33 PM on May 8, 2008
Response by poster: I believe it's 896k. Forgot to mention that as well. I read somewhere than VOIP calls required about 90k each, but we could probably add another DSL line and still save a fortune.
posted by abkadefgee at 7:39 PM on May 8, 2008
posted by abkadefgee at 7:39 PM on May 8, 2008
I used Skype for a bit to dial out, but it's worth noting that the number it shows is generally nonsense, either all 0's, or just random. But I was paying about $3/month for unlimited dialing out. I'd also warn you that I'm yet to find a "Skype handset" that isn't a total piece of junk, though I haven't tried all of them. I just use a normal old headset and plug it in with the Skype client.
This page suggests that it's 3-16 KB/sec, or 24-128 kbps. They claim that Skype will work on a 33.6kbps modem, but obviously not with awesome voice quality. In theory, if you're running out of bandwidth, Skype will sense this and back down on the bitrate. I don't know how smoothly this works in reality, though.
posted by fogster at 8:06 PM on May 8, 2008
This page suggests that it's 3-16 KB/sec, or 24-128 kbps. They claim that Skype will work on a 33.6kbps modem, but obviously not with awesome voice quality. In theory, if you're running out of bandwidth, Skype will sense this and back down on the bitrate. I don't know how smoothly this works in reality, though.
posted by fogster at 8:06 PM on May 8, 2008
You should use a real VoIP solution, i.e. set up asterisk and get a SIP provider. Trust me, you'll make out better in the long run.
Skype is for people who :
1) use laptops under questionable internet situations,
2) need to get older relatives to install it easily, or
3) also need video chat and/or IM.
posted by jeffburdges at 5:25 AM on May 9, 2008
Skype is for people who :
1) use laptops under questionable internet situations,
2) need to get older relatives to install it easily, or
3) also need video chat and/or IM.
posted by jeffburdges at 5:25 AM on May 9, 2008
Just like jeffburdges said, asterisk is one of the most professional solutions for this kind of thing.
You use one server as an asterisk box and use softphones on your standard computers. Or you can buy (rent?) real voip telephones. This way you can connect nearly as many voip telephones as you want to the asterisk box and asterisk makes as many concurrent outgoing calls as you want.
An easily installed linux distribution that is dedicated to asterisk is trixbox community edition. An alternative to that is AsteriskNOW.
Mor information is available in the freely downloadable O'Reilly Book "Asterisk: The Future of Telephony". There are also a lot of tutorials available.
If you dread the setup of your own asterisk server, where you still have to find a voip provider, you might want to find a full service VoIP provider that basically runs an asterisk server for you. This way you just need to connect your dedicated VoIP phones / software phones to them, which might be easier and more reliable for you if this is just a one-off thing. "Real" VoIP is much more reliable and better sounding than skype. You can even set your own callerid.
There is a list of service providers on the excellent voip-info.org site.
There are several codecs available for VoIP calls. Internally (LAN) the codec of choice G.711 is the same codec used with ISDN (T1) phones it has better than analog phones quality and usses 64kbit/s per Call.
The GSM (like the mobile phone standard) codec is another popular free choice which uses less bandwidth 13.2 kbit/s.
Additainall you have to add some overhead for the IP communication so that with GSM you get ca. 30kbit/s per call (in + out + overhead) (bandwidth calculator)
posted by mmkhd at 6:49 AM on May 9, 2008
You use one server as an asterisk box and use softphones on your standard computers. Or you can buy (rent?) real voip telephones. This way you can connect nearly as many voip telephones as you want to the asterisk box and asterisk makes as many concurrent outgoing calls as you want.
An easily installed linux distribution that is dedicated to asterisk is trixbox community edition. An alternative to that is AsteriskNOW.
Mor information is available in the freely downloadable O'Reilly Book "Asterisk: The Future of Telephony". There are also a lot of tutorials available.
If you dread the setup of your own asterisk server, where you still have to find a voip provider, you might want to find a full service VoIP provider that basically runs an asterisk server for you. This way you just need to connect your dedicated VoIP phones / software phones to them, which might be easier and more reliable for you if this is just a one-off thing. "Real" VoIP is much more reliable and better sounding than skype. You can even set your own callerid.
There is a list of service providers on the excellent voip-info.org site.
There are several codecs available for VoIP calls. Internally (LAN) the codec of choice G.711 is the same codec used with ISDN (T1) phones it has better than analog phones quality and usses 64kbit/s per Call.
The GSM (like the mobile phone standard) codec is another popular free choice which uses less bandwidth 13.2 kbit/s.
Additainall you have to add some overhead for the IP communication so that with GSM you get ca. 30kbit/s per call (in + out + overhead) (bandwidth calculator)
posted by mmkhd at 6:49 AM on May 9, 2008
Response by poster: Ok, I just ordered ATA's and VOIP lines from leasedminds.com. We're supposed to be running next week, and I'll try to let you know how it goes. Like I said in the OP, I haven't had any experience with asterisk, and I don't have the time to learn before we get things running. This should be a lot easier.
posted by abkadefgee at 11:13 PM on May 10, 2008
posted by abkadefgee at 11:13 PM on May 10, 2008
This thread is closed to new comments.
posted by winston at 6:33 PM on May 8, 2008