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tonyhughes

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#11485 26-Jan-2007 11:18
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With my Asterisk box DMZ'd (for testing only - not full time!!) I can use IDEFISK IAX softphone over my laptop with Sierra card, and everything works great (call quality and latency is acceptable, but not perfect).

If I use any SIP softphone, I can connect to the PBX just fine, and make/recieve calls, and hear ringing tone, but there is no actual call audio either way once connected.

Anyone else had similar problems? I initially thought it was port issues, but I thought DMZ would remove that as an issue?








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sbiddle
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  #58900 26-Jan-2007 11:54
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A DMZ doesn't get around the issued with NAT. All it does is open all the ports and not solve the actual routing issues that is causing the packets to disappear.

IAX gets around the problems with NAT transversal but unfortunately trying to use SIP devices behind NAT can be messy, especially if you're using NAT at both ends..:-(

Have you tried using a STUN server on the softphone when connecting remotely? This can solve a lot of issues.


tonyhughes

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  #58917 26-Jan-2007 14:48
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Okay, thats a bit clearer, I guess i understood what DMZ achieves, but not how it achieves it. I have a few more tricks to try then.







 
 
 
 


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  #58920 26-Jan-2007 14:55
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Have you followed the setup stuff in Ben's Trixbox without tears guide? Setting your fixed IP or dynamic host in your sip.conf or sip_nat.conf and setting your localnet as appropiate. You should also set NAT=YES in the settings for the extension you want to use as your remote phone.

nat=yes
externip=<your fixed external IP> or
externhost=< myasterisk.freedns.com> (see note)
localnet=192.168.1.0/255.255.255.0
externrefresh=10

tonyhughes

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  #58922 26-Jan-2007 14:58
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No i hadnt tried that. I suspect that that will cure my ills.

Its no biggie, because IDEFISK is nice and light, and thats really what I wanted, though being able to use any SIP phone remotely is an option id like to have.

Id love one of the cordless wifi SIP phones, but they are all up around the $300 - $500 mark still, despite being a cross between a $40 cordless phone and a $40 wifi adapter, with about $10 of extra smarts thrown in :-(







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  #58925 26-Jan-2007 15:15
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tonyhughes: Id love one of the cordless wifi SIP phones, but they are all up around the $300 - $500 mark still, despite being a cross between a $40 cordless phone and a $40 wifi adapter, with about $10 of extra smarts thrown in :-(

Yeah, I looked at them too and came to the same conclusion...

Wait a bit longer and the price is sure to come down once more players enter the market.  At present it's mainly Philips, Logitech, Netgear et al i.e. big name suppliers.  Once the no-name players enter the market, competition will drive the prices down.

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  #58927 26-Jan-2007 15:21
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WiFi SIP phones have so far been a big disappointment. Even Linksys haven't managed to get any momentum going in the marketplace. The biggest issue is that you ideally want a phone that can handle HTTP authentication which means giving it an OS and web browser. If it doesn't you can't use the phone at most public hotspots.

tonyhughes

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  #58933 26-Jan-2007 15:55
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sbiddle: WiFi SIP phones have so far been a big disappointment. Even Linksys haven't managed to get any momentum going in the marketplace. The biggest issue is that you ideally want a phone that can handle HTTP authentication which means giving it an OS and web browser. If it doesn't you can't use the phone at most public hotspots.

Agreed. It would be great to just be able to preset settings for home, work, Telecom hotspots, and other services, and just have the unit automatically connect to any available service. If it could do that, id be tempted at $300.







 
 
 
 


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  #58935 26-Jan-2007 16:44
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tonyhughes: It would be great to just be able to preset settings for home, work, Telecom hotspots, and other services, and just have the unit automatically connect to any available service. If it could do that, id be tempted at $300.

It would also need to be smart enough to handle disconnection so the customer doesn't keep getting charged for Hotspot time after the call has ended.

For example: At a Telecom Hotspot, you have to go back to the pop-up window and click "Disconnect" once you have finished with the service. Or else, you keep getting charged for the time @ $x per hour.

I can see it getting tricky to have a phone which caters for all the different billing systems used at Hotspots. So far as I've found, they are all different. Some standards need to be put in place to make the WiFi SIP phone a viable idea.

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  #58936 26-Jan-2007 17:17
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I believe the Telecom system does have a timeout on it, but I guess you wouldnt want to rely on it...







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  #58951 26-Jan-2007 18:54
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The only phone that does do HTTP authentication is Linksys WIP330 - and that's because it runs on Windows CE and has a browser onboard.

richms
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  #59021 27-Jan-2007 14:41
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If hotspots would drop all the http auth crap and just use 802.1x to authenticate then this really would be a non-issue, but I guess they like the idea of having advertising on a web interface that people have to look at to use it. 




Richard rich.ms

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  #59026 27-Jan-2007 17:14

richms: isn't 802.1x is over-the-top for hotspots?
I mean there are WPA2-PSK hotspots (chillispot) but the main idea behind HTTP captive-portal authentication is to have zero-config for clients - not advertising.

oh and to be on-topic SIP+NAT != FUN Smile




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  #59038 27-Jan-2007 21:29
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barf: oh and to be on-topic SIP+NAT != FUN Smile

Yes indeed, we drifted off topic a bit there didn't we...  some interesting points made though.

WxC seem to have SIP + NAT well and truly sorted with VFX, at least so far as Linksys devices are concerned.

One of the techniques used seems to be a STUN server which I tried to get working when using my PAP2T with iTalk.  However, the server end also needs to be configured correctly which it obviously wasn't in iTalk's case, hence we ended up not receiving a lot of calls, or in other cases, the caller couldn't hear us even though we could hear them.  Obviously there were some issues there with NAT traversal of the RTP packets.

It must have taken quite a lot of trial and error to get this right in WxC's case and I can understand them not wanting to divulge exactly how it's done.

Hopefully Tony has managed to get his SIP softphone audio packets traversing NAT OK by now.

How's it going Tony?

tonyhughes

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  #59040 27-Jan-2007 22:00
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I have had some other things take priority over this issue (seeing as IAX is working fine remotely in the meantime).

I just moved my servers and WAG54GP2V2 to the garage (i have a telephone jack out there), and have a wireless bridge set up to a second seperate wired lan in the house. The PAP2T is on that 2nd LAN, and both Asterisk+SIP services, and VFX are working perfectly over that setup, with no extra perceptible latency introduced.

The physical route to my PAP2T is currently as follows:

WAG54GP2V2 [wifi] Belkin Wireless Bridge [eth] Netgear WGR614 [eth] PAP2T

My laptop is currently connected as:

WAG54GP2V2 [wifi] Belkin Wireless Bridge [eth] Netgear WGR614 [wifi] Laptop

My ping time to the WAG54 is 3ms average.
My ping time to the Wireless Bridge is 2ms average.
My ping time to the PAP2T is 1ms average.

(O.T. but perhaps interesting).

The wireless bridge / netgear router setup may seem a bit odd, but its what I had available to me...

Of course the Asterisk box is still directly connected to the WAG, so I havent introduced any complexity there.







richms
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  #59045 28-Jan-2007 01:02
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Careful with those wireless bridges, I tried to use one once, and it did some mac address spoofing to make everything on the ethernet side of it appear as one mac address to the accesspoint and anything on the accesspoint side of the bridge. This means that if your laptop or something roams between downstream of the bridge, over to the main accesspoint, it totally loses connectivity with stuff on the bridge untill it is powercycled.

Also broadcast stuff like uPNP media storeage did not work on the bridge.  




Richard rich.ms

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