[VoIP] I am now online!

[David Josephson]

Steph,

We are migrating from the always-wired SDM/FDM/TDM telephone network 
that concentrates on making a single path reliable, to a web paradigm 
that concentrates on making many unreliable paths available with the 
understanding that most of the bits will get through most of the time. 
The short answer is that each link is unreliable. The system works fine 
when multiple paths are available to a cooperative server. There are 
lots of random wifi hotspots. In my experience, about half of them are 
open and pass the necessary VoIP port traffic. Port forwarding isn't 
needed, only pass-through. Mobile roaming is not well addressed by the 
802.11 protocols used by WiFi; it can take a few tens of seconds to get 
set up on a new hotspot. Newer protocols address this more intelligently.

But to answer the original question, you can set the interval after 
which your WiFi phone registers with its host. One to ten minutes is 
typical. Likewise you can set your SIP server to time out after a 
minute, forcing all the connected hosts to re-register.

I have one of the first WiFi VoIP SIP phones, the Pulver "WiSIP" aka 
Zyxel 2000. It works if you keep it working with a network that uses the 
same SSID for all of its access points. Anything else and it's stupid. I 
am going to try the new Linksys WIP-330 which seems to  have a few more 
cycles of feature refinement.
> Hmmm....  interesting.  Thanks.
>
> My question in part comes out of wondering how wi-fi phones can work 
> reliably when one walks within range of some random wi-fi hotspot that 
> the phone owner has no control over and which won't necessarily have 
> port forwarding enabled for special ports that might be required for 
> Voip service.
>
> Steph
>
> Chad Perkins wrote:
>   
>> Yes and no.  If you have your Asterisk system "register" with another gateway/server 
>> the registration process takes care of the port forwarding and "keep alive", but that 
>> means a central server.  That's not how CNET is setup; in CNET we use ENUM to 
>> find each other and the tandems are all "peers" without a central server.  No central 
>> server, no central point of failure.
>>
>> Now I used to "register" a second non-CNET Asterisk (off-site) with my main one.  I 
>> did it for other reasons, but it would accomplish what you suggest albeit with some 
>> additional complication and an additional trip across the Internet for the voice stream 
>> (this is essentially the same configuration as the ATA method except the terminal 
>> device is another Asterisk).
>>
>> Chad
>>
>> On 23 Feb 2008 at 11:57, Steph Kerman wrote:
>>   
>>     
>>> Is there some way to add something to the * code or config files to
>>> keep the connection alive by placing dummy test calls once per minute?
>>> =SK=
>>>     
>>>       
>>   
>>     
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