Batboard

Hello all. I'm just currious how many batlabbers use VoIP phones. The reason I ask is that I use VoIP in everyday work and as a result, we have an Asterisk server running.

There are several *free* VoIP applications that will let people use their PC as a VoIP phone out there and if interest is great enough, I might consider setting up a VoIP server so batlabbers could contact each other via VoIP.

Word of warning: If you're on a modem, you need not apply. You're going to need 2-channel ISDN at the MINIMUM.

John

[I'm making this thread a sticky in hopes that other people will get on the bandwagon. Also, if it's not here for people to see, then it'll never expand. -Alex]

I'd be interested in any project like that, I have extensive VoIP expereince.

I have a Cisco CallManager system at home with a PSTN gateway and IP phones, I have a few friends who also use the system remotely as well.

I've also run Asterisk, it's a great system, considering the cost, it's such an amazing system.

It's a great technology, and it's mainstream now, anyone who thinks it's not ready for mainstream needs to look again (and talk to someone who knows what they are talking about) I have legal firms, and govt agencies running them.


-- Grint.

Drop me a PM with some details (authentication PW, preferred extension, etc) and I'll set you up on our * server and we can do some test runs.

John

VoIP is huge...

Niagara Regional Police are currently replacing their entire phone network with VoIP..the plan is eventually for all Regional Niagara services to be VoIP.

I've used a Cisco phone, the audio quality is (in my opinion), better than a regular ePOTS phone.

I'd be all for something like that!

-Mike

I have a Mitel 5055, and I'm on the FWD, IPTel and FreeIPCall networks.

FWD#: 401016
IPTEL#: 91991
FIPC#: I can't remember, it's some 10-digit crap

I'd love to test the Mitel 5055 with other services, such as CallManager and/or Asterisk... PM me if you want to add me

BTW does anyone know where I could get some Mitel 5055's for cheap? They don't show up on eBay very often, but they're such great phones...

I have always wanted to explore VoIP, but don't have any equipment. Now if there is some freeware available or what not, I'd be all for something like you are speaking of. Keep us updated!

David

Does anyone know how I may be able to run VoIP behind my university's firewall? I used to use FWD... but it doesn't work behind our firewall here, I guess. And I doubt the university will open up some ports just for me.

I think a Batlabs VoIP network would be awesome!

I'm on board.

What do I need to get started.



-Alex

Ok the biggest problem we are going to have is NAT.

For those who know IP well, VoIP protocols use ip info in the payload, so the IP of the phone is in the payload, which is a problem, because when things get natted, that address doesn't change.

I would be willing to host an H.323 site, as my router can fix those payload issues (Cisco), there is no cheap firewall out there that can.

SIP is not supported until CallManager 4.0 (a few months away) but we can do H.323 and SCCP (Skinny)

This just might work. The other option is for everyone to connecto to FWD (free world dialup) if you can't connect to FWD, then you have firewall issues of your own, and nothing will fix that.

If you have a Linksys or Netgear router, the DMZ feature will help fix your issues, the UDP packets don't ususally penetrate most NAT/Firewalls so if your system isn't working set the DMZ to the IP Phone / PC

grinthock wrote:Ok the biggest problem we are going to have is NAT.

For those who know IP well, VoIP protocols use ip info in the payload, so the IP of the phone is in the payload, which is a problem, because when things get natted, that address doesn't change.

I would be willing to host an H.323 site, as my router can fix those payload issues (Cisco), there is no cheap firewall out there that can.

SIP is not supported until CallManager 4.0 (a few months away) but we can do H.323 and SCCP (Skinny)

This just might work. The other option is for everyone to connecto to FWD (free world dialup) if you can't connect to FWD, then you have firewall issues of your own, and nothing will fix that.

If you have a Linksys or Netgear router, the DMZ feature will help fix your issues, the UDP packets don't ususally penetrate most NAT/Firewalls so if your system isn't working set the DMZ to the IP Phone / PC

Uh huh - what he said!



Can someone explain this whole deal to me, who has never messed with VoIP before?

Or even easier - whore yourself a 2nd external IP from your provider, like I did

I forgot to note previously: My FreeIPCall number is (131) 229-1445.

You can run many SIP applications from behind a NAT without any problems. I've got two SIP phones running behind a NAT here at my house right now without any issues.

OK. If you don't own any SIP hardware, you can run the following under Windows, Mac or even under linux using WINE (what I'm doing for my Softphone) http://www.xten.com/index.php?menu=prod ... u=download

You want X-Lite

I'm going to beta with a few batlabbers who have already contacted me and we'll come up with a decent dialplan and then take on the masses.

If you want HARDWARE, you can get two SIP adapters that you then plug a regular phone into at http://www.sipphone.com.

I would try the software first though. It works fine on most computers and it will give you a chance to play without spending any money. If there is enough of an interest, I'll see what I can do about getting us a bulk discount on some decent dedicated IP phones. We got the price down to aout $65.00 each during the last rollout we did on another project.

OK. I'll be contacting a few members and we'll keep you all posted with the progress of the BAT-phone project.

John

John,

Put me on the list.

73

I got it running using X-Pro, and I'm trying to set it up using my Mitel hardphone. I will post X-Lite/X-Pro configuration info later. Works great for now!

k4wtf wrote:You can run many SIP applications from behind a NAT without any problems. I've got two SIP phones running behind a NAT here at my house right now without any issues.

This is not quite right. The router/gateway that is performing NAT must be SIP aware, as the IP address is in several places in the SIP packet itself, and also must be translated. Cisco routers running a recent IOS (12.2(11)T I think and up, and 12.3) can do it. As can D-Link DSL-504s I believe.

Oh BTW... I can host a SIP server here if people like.

We have our own SIP product that is quite successful, and so setting up a server shouldn't be much of an issue at all.

OK. I downloaded that X-Ten thing. Now what do I do?

nmfire10 wrote:OK. I downloaded that X-Ten thing. Now what do I do?

You need to associate to a SIP server...

Give me a bit and I'll set one up.

k4wtf wrote:I'm going to beta with a few batlabbers who have already contacted me and we'll come up with a decent dialplan and then take on the masses.

Hey John,

Have you got a SIP server running for this, or are you just targeting phones directly using a dialplan?

Exactly, SIP Aware router is key, unless you DMZ to your IP Tel device.

If we get up and running, i'll put the SIP load on one of my IP Phones here and join up.

grinthock wrote:Exactly, SIP Aware router is key, unless you DMZ to your IP Tel device.

Or just plain aren't running NAT

OK. Here's the deal. I'm up and running with one of the other batlabbers. I'm behind a NAT with a NON-SIP AWARE NAT device and he's in the clear. For those of you who don't use SIP very often, there is such a thing as a STUN server that will assist you if your NAT/firewall is hosing you.

And yes, I am running a SIP server. I created a context on our inhouse SIP server for use as the "BAT-line"....

All "BAT-line" numbers will be BAT-nnnn as in: 228-8888 (which happens to be me)...

If we outgrow 9,999 members using the bat-line, I figure we'll deal with it then.

I don't need anyone else to host anything.

Now, for those of you who have downloaded the soft-phone or have hard-phone devices at your disposal, let me know what you would like your number to be and I can get you added and send you the appropriate configuration information to authenticate to our SIP server.

John

Don't matter to me what the number is. 1148 would be easy for me to remember if that is possible. I still have no idea how to set any of this up in the program settings.

k4wtf wrote:I don't need anyone else to host anything.

OK, cool... Must have missed that bit I won't bother setting one up then

k4wtf wrote: Word of warning: If you're on a modem, you need not apply. You're going to need 2-channel ISDN at the MINIMUM.

2-channel ISDN, what the hell is that?

Why didn't you just say fractional T-1

ASTRO_25 wrote:

k4wtf wrote: Word of warning: If you're on a modem, you need not apply. You're going to need 2-channel ISDN at the MINIMUM.

2-channel ISDN, what the hell is that?

Why didn't you just say fractional T-1

Because it's not a fractional T-1

2 channel ISDN is Basic Rate ISDN (BRI). Different layer 1.

Just being sarcastic... that's all.

Could we have just said 128K?

ASTRO_25 wrote:Just being sarcastic... that's all.

Could we have just said 128K?

Doh! Never know... some people might not know about it

Oh, and for what it's worth, I've on many occasions successfully had VoIP calls running with G729 codec across a 33.6 modem (SIP signalling) with no problems

G729 is great if you're (1) willing to deal with the poor audio quality and (2) pay for a license to use it on your SIP devices. I have a 2-channel license for our SIP server because I have two devices that sit on low bandwidth links but, since I only have those 2 channels for G729 and they're obviously for something "real" and not the "Bat-phone", I said you'll need the bandwidth to support the other CODECs.

John

k4wtf wrote:G729 is great if you're (1) willing to deal with the poor audio quality and (2) pay for a license to use it on your SIP devices. I have a 2-channel license for our SIP server because I have two devices that sit on low bandwidth links but, since I only have those 2 channels for G729 and they're obviously for something "real" and not the "Bat-phone", I said you'll need the bandwidth to support the other CODECs.

John

Hmm.. Fair enough. Our Cisco phones have G729 built in, so they must be covering the license fee. Strikes me that licensing codecs should be the responsibility of the phone and not the SIP server

You're confusing a cisco "Call manager" with a SIP server. Two different animals man. A sip "server" is actually terminates the calls, IE; each party connects to the sip server and it bridges them together. Thus, the server has to be able to speak whichever CODEC the phone wants to speak.

John

k4wtf wrote:You're confusing a cisco "Call manager" with a SIP server. Two different animals man. A sip "server" is actually terminates the calls, IE; each party connects to the sip server and it bridges them together. Thus, the server has to be able to speak whichever CODEC the phone wants to speak.

John

Incorrect. A SIP server acts purely as a proxy between different SIP User Agents (the phones). A SIP server by itself does not take part in the RTP side of things at all.

A SIP conversation goes something like this:

1) User Agent sends an INVITE packet with the destination to the SIP server/proxy.
2) Server/proxy (it's actually a proxy so I'll call it that from now on) sends back a "Trying" to the User Agent.
3) Proxy checks to see if it knows how to get to the target (either locally registered or if needed by DNS lookup) and forwards the INVITE to the destination.
4) Destination sends back a "Trying" to the proxy
5) If destination is not busy, destination sends back a "Ringing"
6) Proxy forwards Ringing back to calling user agent
7) When the phone is picked up, destination sends an "OK" to the proxy
Proxy forwards to calling User Agent
9) The initial invite and the OK packets also contain codec and RTP stream settings, importantly which IP and port to connect to for the RTP stream. The destination selects what it wants to use from the INVITE settings and sends it back to the caller. If the caller agrees, at this point it sends an ACK to the proxy, and the proxy forwards to the destination.
10) The RTP stream is then established between the caller and destination directly, without involvement by the SIP proxy.

Any SIP server that does get involved in a stream is generally doing it for conference call purposes, or it's a hack to get around dodgy SIP UA problems

OK,

I setup sipphone...

now what?

-Alex

alex wrote:OK,

I setup sipphone...

now what?

-Alex

Get an IP address, SIP UID and maybe also username and password from John, register to his SIP server then place a call to someone else who's also on

If someone (John?) gets me some details, then I'll register my SIP phone to it and someone can call me

Sundown is very correct. SIP and CallManager and very different, CallManager acts as a call control agent, where a SIP server can act as a proxy as well.

Going with G.729 or GSM over say G.711 is not necessarily going to help you with sound. The 2 biggest problems with IP Tel calls are Jitter, and Packet Loss, RESPONSE TIME means more.

If I say lose 200 packets during a conversation, I don't even notice (my IP phone connecting to my office for instance over a VPN) I run G.711, which is 80K/sec codec, as big as it gets, and it runs fine. G.729 just sounds poor by comparison. Response time means everything, more importantly, a "Constant" connection is even better, if your response times jumps from 20ms to 500ms, that confuses the phone, it is trying to buffer and adjust forthe difference in delay.

Bottom line, don't worry too much about your codec, if your connection stinks, changing the codec probably won't help that much.

CallManager does have the ability to proxy and transcode, but not with SIP. It only supports H.323 and SCCP (Skinny) as of now...

Oh well, SIP is a better solution for us for now

grinthock wrote:Going with G.729 or GSM over say G.711 is not necessarily going to help you with sound. The 2 biggest problems with IP Tel calls are Jitter, and Packet Loss, RESPONSE TIME means more.

I agree. But if you remember, I was running a call across a 33.6 modem, which obviously can't support the 80k required for G711 G729 needs about 24k of bandwidth, which can reasonably comfortably fit inside that size pipe.

Running RTP header compression can reduce a G729 call bandwidth requirement down to 13/14k, allowing more calls in a low bandwidth link environment (to reduce Telco costs).

Very definately important to ensure reasonably predictable packet delivery to reduce jitter though.

Okay, VoIP isn't exactly my strength.

What do I need to download to get onboard?

-Mike

OK, i'm pissed off and annoyed at this stupid software client.

I told my firewall to allow SIP connections, opened the ports properly...

What gives.

I am trying to talk to Jaymz and PJ now - I have a # but it still is telling me it timed out.

WTF.

-Alex

alex wrote:OK, i'm pissed off and annoyed at this stupid software client.

I told my firewall to allow SIP connections, opened the ports properly...

What gives.

I am trying to talk to Jaymz and PJ now - I have a # but it still is telling me it timed out.

WTF.

-Alex

In addition to allowing UDP 5060, you also need to allow the UDP ports used for RTP. See if you can lock the UDP port to a specific one in the client, or find out what range it's going to use, and open that as well.

Sundown wrote:

alex wrote:OK, i'm pissed off and annoyed at this stupid software client.

I told my firewall to allow SIP connections, opened the ports properly...

What gives.

I am trying to talk to Jaymz and PJ now - I have a # but it still is telling me it timed out.

WTF.

-Alex

In addition to allowing UDP 5060, you also need to allow the UDP ports used for RTP. See if you can lock the UDP port to a specific one in the client, or find out what range it's going to use, and open that as well.

Hmm...

Q: Will the SIP phone work behind a firewall?

A: If you have a firewall, you may experience problems with your SIP phone. However, NAT firewalls should not cause any problems. Please have you systems administrator open ports range 5004 and 5060-65534 UDP in your firewall to allow SIPphone calls.

I'm using the client at http://www.sipphone.com... just to see if it works.

-Alex

I'll try opening the other one you specified and see if that works. 5060-65534 is REALLY a broad range

Alex:

Are you able to make the default connections for the client? (i.e.) **, *0, 411, or any 800 number?

I've got it doing all of the above, but haven't tried to talk with another live user yet.

Yup

Mine does all the above - I can call the #'s you mentioned - no problem.

Just keeps failing...

-Alex

Hey I have an extra 7910 and and extra 7960.

the 7910, I beleive only does callmanager, however the 7960 can do SIP, SCCP, and MGCP.

I am new to VoIP, at work we have fully transitioned over to it. All that I can say is WOW! The voice quality is definitely better the POTS. Not only that, but all the features our service provider includes in basic service is really cool.

I am interseted in this. I try and load a SIP image into the 7960.

What version should I get?

OK folks... Here is what we have so far:

2285329 jaymz
2280896 kg6eaq
2283223 humhead
2284162 cpd534
2281414 fireradio
2289111 Alex
2280880 The Pager Geek
2280909 911radio
2285260 Johnny Grep
2280000 & 2288888 = Me (k4wtf)

If you're on this list and you don't yet have a password, send me a PM with the password you want to use and I'll reply with your complete config instructions.

If you are not on this list and want to be part of the Bat-Phone project, send me a PM with the 4-digit prefix you want as well as the password you want to use and I'll get you set up as well.

Johnny Grep was going to be working on "simple" instructions for people to get set up but, I don't know if he's done that yet.

Johnny asked me last night if it would be possible to set up a conference bridge. It *is* possible but, I don't have the appropriate hardware to do it on our Asterisk server. It requires a Zaptel. If the demand is high enough, I'll see what I can do to get a card and get us conference bridge capability.

Oh, and if you tried to send me a PM and it didn't go through, it was because my inbox was full. I just cleaned it out. Sorry.


John

A couple of test numbers have been set up.

BAT-TEST (228-8378) will go to a humerous recorded message so you can test your bat-phone setup without actually boring someone.

BAT-ECHO (228-3246) will put you into an "echo test" that will echo everything you say back to you. This will let you know how well your connection and SIP hardware/software are working.

Enjoy...

John

OK. Here is how you configure X-Lite for to connect to the BAT-LINE.

1. Launch the program
2. Just above the "3" key on the telephone keypad in the program, you'll see a button that looks like a notepad or something. This is the menu button. Click on that button.
3. Double click on "system settings"
4. Double click on "Sip proxy"
5. You'll see something like "[Default]" Double click on it.
6. Set "Enabled" to YES
7. Set Display name to Whatever but, please include your BAT-nnnn number.
8. Set username to your batphone number, IE; "228nnnn"
9. Set Authorization User to your batphone number.
10. Set password to the password you have requested.
11. Set Domain/Realm to "BAT-PHONE"
12. Set Sip Proxy to "sipproxy.enterzone.net:5060"
13. Set Use Outbound Proxy to NEVER
14. Set Send Internal IP to NEVER
15. Set Register to ALWAYS
16. Close out the menu (hit the X at top right)
17. Close out of the program
18. Restart the program

You should now be connected and authenticated to our SIP server. If all is well, you should be able to dial BAT-TEST (just type the words and hit enter) and be connected to the test.

If you have problems, post them to the thread and we'll try to get you working. I do not have any problems authenticating via NAT through my wireless router (Linksys BEFW11S4) so I know it is possible to get up and running.

There is a list of firewalls that are known to work/not work at http://sipphone.com/routers/

Take what the list says with a grain of salt though if it says your router/firewall won't work. I see a few listed as "doesn't work" that I know happen to work fine.

I am not a SIP or VoIP expert but, I have managed to get over 50 phones working with this system from various locations so, hopefully we'll be able to get YOU working too!

John
[HumHead: Information updated for K4WTF 8/26/04]

For those of you who would like an INEXPENSIVE "hard phone" for use with SIP networks, here is a link to an auction that has the phone I use.

http://cgi.ebay.com/ws/eBayISAPI.dll?Vi ... gory=11909

It's not a Cisco 7960 but, then again, it costs less than 1/10...

John

For those of you with sonicwalls - they should work without an issue.

-Alex

I seem to be having some sort of problems. I'm not familliar with VoIP so I have no idea what it is. I beleive I entered everything correctly but it keeps timing out and won't connect to the server. When I open X-Lite it says "Logging In" for about a minute then it says "Login Timed Out Contact Systm Admin"

I've got McAfee Security Center if that matters

Any ideas?

OK folks. I found an issue on our side of things.

Use the following address as the SIP proxy:

sipproxy.enterzone.net:5060

For some reason, the software was getting confused when I was binding it to one of the other IP addresses on the box so, I had to go for the primary OC3 interface of the box. I'm not sure why but, try with the IP above as the "SIP Proxy" and you should be fine.

John

[HumHead: Information updated for K4WTF 8/26/04]