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Unread 10-27-2010, 09:10 AM   #7
emoci
 
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As far as I understand....

With @voxalot.com ...the associated A adress (IP) is actually the webserver. Which means if te calling end does not do an SRV lookup ... the call is unlikely to get anywhere ... it is by default hitting a web server ...

With @xx.voxalot.com... the associated A adress (IP) corresponds to one of the actual proxies (I believe it is proxy 1 in each of the us and eu). That means even if the calling party does not do well with SRV lookups...the call will end up in a server that is voice-aware. (there could still issues if for some reason one of the proxies is not working and SRV lookup is not being done...because the failover mechanism would not be able to compensate automatically).

From my experience: Most SipPhones, ATAs, SoftPhones when being used to make direct SIP Calls do pretty well with DNS SRV. Doing DNS SRV lookups when it comes to registration to a specific server seems to be a beast of its own though (so those same software/devices can do well on the first but fail miserably on the second).

When you're actually forwarding a call from OpenSips or Sip2Sip ... then the SRV lookup rests with the provider itself rather than the SoftPhone.

One alternative to overcome SRV issues (especially when forwarding from a third party provider) is to try forwarding to *010123456@sipbroker.com (where 123456 is your VoXalot number). The SipBroker.com server is actually both web and voice on the same server... and SipBroker itself is pretty good at doing SRV lookups for failover situations from there...
(I personally tend to avoid this unless it is an extreme case, I feel like it adds some unneccessary extra latency--but I must admit this assumption of extra latency is based on no real proof on my end except for a gut feeling--)

If you do put a setup guide together...let me know...I'd be happy to add it to the Wiki...
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