VoXaLot Launches European Server

[krbonne]

Gegroet,

Originally Posted by Tony

To all VoXaLot Members,
On the 18th of January we posted a thread on our support forum stating that we would be building a European cluster to improve the quality of the VoXaLot service for European members. This decision was based on the feedback we received from our European members in a recent survey we conducted.

Great!


Just some questions. (I'm not such a big expert in SIP).

1/ Does logging in on this server instead of the US server affect the SIP URI. Is it still "SIP:number@voxalot.com" or has it become "SIP:number@eu.voxalot.com"?


2/ As you say that the server is designed to be "fully operational" for SIP-calls between voxalot-users; why are there different SIP-broker id for voxalot-servers in the US, AU and EU?

Can you explain the difference in set-up between these two kinds of kinds of connections (inside the voxalot domain and when using SIPbroker)?



Many thanks in advance,
Cheerio! Kr. Bonne.

[code-]

Good ping in Norway, echo test is much more responive now

Pinging eu.voxalot.com time=39ms
Pinging voxalot.com time=117ms


Those of you complaining about ping time should post your location as well...

[martin]

Originally Posted by krbonne

Just some questions. (I'm not such a big expert in SIP).

1/ Does logging in on this server instead of the US server affect the SIP URI. Is it still "SIP:number@voxalot.com" or has it become "SIP:number@eu.voxalot.com"?

You can use both. The eu.voxalot.com domain is a more direct path though. As the clusters fully inter operate with each other any domain name will find the final destination.

Originally Posted by krbonne

2/ As you say that the server is designed to be "fully operational" for SIP-calls between voxalot-users; why are there different SIP-broker id for voxalot-servers in the US, AU and EU?

Once again we are providing direct access codes. Like the SIP URI's any SIP-Code can also be used, however using the most direct code will reach its destination a little quicker.

Originally Posted by krbonne

Can you explain the difference in set-up between these two kinds of kinds of connections (inside the voxalot domain and when using SIPbroker)?

I hope the above 2 answers address your question. In a nutshell any URI or SIP-Code can be used to reach any VoXaLot member. The only difference is that the more direct path will connect to the member a little quicker.

[krbonne]

Gegroet,


First of all, thank you for replying. I'm still learning about how SIP really works so please excuse me additional questions.

Originally Posted by martin

You can use both. The eu.voxalot.com domain is a more direct path though. As the clusters fully inter operate with each other any domain name will find the final destination.
...
I hope the above 2 answers address your question. In a nutshell any URI or SIP-Code can be used to reach any VoXaLot member. The only difference is that the more direct path will connect to the member a little quicker.

[/quote]


What exactly do you mean with "a more direct path"?

I did some tests (I like to find out things myself before annoying other people with questions :-)) where I use the SIP-client on my laptop and run "tcpdump" at the same moment to verify what really happens.

The I made a PSTN-call to the SIP dail-in number here in Belgium (packetnet.be) and made a call to *010<my-number> and *031<my-number>

In both cases, I got the same thing:
- a SIP-message from eu.voxalot.com telling me that there is an incoming call
- an outgoing UDP-flow from my client to packet.be for the voice-call.

Now, of course, I cannot see what happened earlier. I don't know what happened in the earlier stages when I made the call to *010...

But can I assume that the difference is only in the "control" part (i.e. the exchange of packets to do the call-setup); but that for the call itself it does not make any difference at all.


Is the situation any different if I would configure my client to use "strickt outbound proxy"?


Again, I'm sorry for the question. I'm still trying to really understand how SIP works "under the hood".


Cheerio! Kr. Bonne.

[martin]

Originally Posted by krbonne

But can I assume that the difference is only in the "control" part (i.e. the exchange of packets to do the call-setup); but that for the call itself it does not make any difference at all.

Yes that is right. It is just the call-setup that benefits. Once the RTP stream is setup it will be the same.

[pianoquintet]

It seems to work differently in my case. I did a tcpdump during both inbound (PSTN to SIP) and outbound (SIP to PSTN) calls and the RTP flow seemed to go through "hosted-by.leaseweb.com" (which is the European Voxalot server) instead of my Italian SIP provider (Messagenet.it) servers.

These were my settings at the time of the experiment:

- my internet provider is Fastweb.it, which puts all its users behind a NAT;
- my SIP client is registered with Voxalot only; STUN is set to stun.voxalot.com.au;
- Messagenet.it, my Italian SIP provider is entered in Voxalot's "Providers" list and both "Active" and "Register" are set to "Yes";
- my Voxalot dial plans point to Messagenet for the relevant outgoing calls;
- I dialled PSTN numbers from my SIP client to test outbound calls and dialled the PSTN number associated with my Messagenet account from a landline and answered them with my SIP client to test inbound calls.

In both cases UDP traffic seemed to go through Voxalot. I would like to have my voice traffic go through the relevant SIP proider instead, both in order to avoid burdening Voxalot's servers and to reduce latency. Can you help?

Thank you,

Carlo

[vickrm]

Hi Martin, today I have migrated all my providers from us.voxalot.com to eu.voxalot.com and I have changed proxy address in my linksys ATA but some registered numbers doesn't ring.
Please can you verify if all my providers are correctly migrated on eu.voxalot.com ? My userid is 902830.
Thanks and continue like this - very good service.
Riccio Vittorio

[martin]

Hello Riccio,

Your providers and migrated settings look very good.

My only thought is that allow an hour or so for your providers to remove the old settings from their cache.

[vickrm]

Originally Posted by martin

Hello Riccio,

Your providers and migrated settings look very good.

My only thought is that allow an hour or so for your providers to remove the old settings from their cache.

Hi Marting thanks for answering, but now that I know that my providers and migrated settings look very good, I think that the problem is in the voicemail. I'm going to explain: When I dial 003906452215271 or 00390690283036 or 00441372422992 start the voice mail at the second ring - that I had to disabled it to receive calls. With the others numbers registered all is ok. Can you help me to find a solution? Bye Vittorio

[martin]

Originally Posted by piscicelli

In both cases UDP traffic seemed to go through Voxalot. I would like to have my voice traffic go through the relevant SIP proider instead, both in order to avoid burdening Voxalot's servers and to reduce latency. Can you help?

Thank you,

Carlo

You seem to have all of the right settings to bypass the VoXaLot NAT handing at the server.

I think we will need to set aside some time to reproduce your test in our test environment.

I have added this as an item in our bugzilla database.