Voxalot / SIP Broker Support Forums

Voxalot / SIP Broker Support Forums (https://forum.sipbroker.com/index.php)
-   Voxalot Support (https://forum.sipbroker.com/forumdisplay.php?f=4)
-   -   VoXaLot Launches European Server (https://forum.sipbroker.com/showthread.php?t=970)

Tony 02-02-2007 02:25 PM

VoXaLot Launches European Server
 
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.

Today I am pleased to announce that VoXaLot has publicly launched its first European server. Based on some early testing that members have conducted, the results show much improved latency times.

The Netherlands ~ 10ms
UK ~ 55ms
Belgium ~ 21 ms
Ukraine ~ 93 ms
Russia ~ 71ms
Norway ~ 32ms

By utilising the new server, our European Members will experience an even greater level of service from VoXaLot. The new cluster has been designed to fully interoperate with the other VoXaLot clusters in the U.S. and Australia, thus ensuring members can reach any other member irrespective of their location.

To use the new cluster, simply change your device / phone proxy to use:

eu.voxalot.com

The domain / realm should be set to:

voxalot.com

Then log into the VoXaLot web site and change your preferred cluster to:

eu.voxalot.com

Note You will need to set the SIP Register flag for each of your providers to "No" before the system will allow you to change your preferred cluster.

To test the change dial: *600 (echo test).

We have also allocated a new SIP Broker SIP-Code *031 that points directly to this cluster. This SIP-Code will also allow anyone to call our European members by dialling *031xxxxxx (where xxxxxx is your VoXaLot number) from any fixed line or mobile phone via the SIP Broker PSTN access numbers listed HERE

We would like to take this opportunity to thank you all for your continued support of VoXaLot. At VoXaLot the Team’s goal here is to ensure our members have the best possible user experience and, where possible, we try and respond to your requests as soon as we can. Please continue to provide feedback on current and possible future functionality.

Tony.

pianoquintet 02-02-2007 04:41 PM

Thank you for this improvement! There is something I am missing, however: if any user can bypass the Voxalot RTP proxy by using a STUN server, why do you bother setting up additional servers closer to your users? Is it simply to make the SIP negotiation faster, which requires Voxalot as the middleman? Or are there cases in which users cannot avoid going through Voxalot for the actual RTP voice traffic and you are thus attempting to reduce voice latency for them?

martin 02-02-2007 07:37 PM

Many of our members rely on server side NAT handling and RTP processing. This new server will help improve their experience and improve call quality.

In addition, VoXaLot initiates Web / Mobile Callbacks from the members preferred cluster. With callbacks, VoXaLot handles the RTP traffic for both call legs. European members that use Web / Mobile Callback should notice significant improvements with this new server.

244751 02-03-2007 01:59 AM

Might be a good idea to make this post a sticky for a while or create a new folder for new enhancement / important annoncements

blanu 02-03-2007 06:37 AM

Quote:

Originally Posted by Tony (Post 5164)
Today I am pleased to announce that VoXaLot has publicly launched its first European server.

Great! You guys rock! Thanks!

// Blanu

zlotowinfo 02-03-2007 09:18 AM

poland: what the difference? my mistake
 
ping to voxalot.com - 51 ms
ping to eu.voxalot.com - 53 ms

so what the difference using other serv to call?

my mistake i write wrong number
ping to voxalot.com - 145-154 ms
ping to eu.voxalot.com - 53-60 ms

but i havent problem with calling with this 150ms

ptruman 02-03-2007 10:02 AM

Doesn't work for me! I deregistered my providers, changed proxy, reregistered them, and changed my ATA.

12 hours later, still failing reg! Switched back to US and it's instantly on :)

martin 02-03-2007 10:39 AM

Quote:

Originally Posted by ptruman (Post 5183)
Doesn't work for me! I deregistered my providers, changed proxy, reregistered them, and changed my ATA.

12 hours later, still failing reg! Switched back to US and it's instantly on :)

Please PM your account number and I will look at your settings

Nakkoush 02-03-2007 01:32 PM

Quote:

Originally Posted by zlotowinfo (Post 5182)
ping to voxalot.com - 51 ms
ping to eu.voxalot.com - 53 ms

so what the difference using other serv to call?

The same here, it looks after pinging eu.voxalot.com that it is slower than us.voxalot.com...

But i will try it later on anyway...

Well done Voxalot!!

Mesmer 02-03-2007 03:24 PM

Quote:

Originally Posted by Nakkoush (Post 5189)
The same here, it looks after pinging eu.voxalot.com that it is slower than us.voxalot.com...

But i will try it later on anyway...

Well done Voxalot!!

It's working for me in the UK. :)

The changed routing makes quite a difference:

US server cluster:

Approximate round trip times in milli-seconds:
Minimum = 87ms, Maximum = 88ms, Average = 87ms

NL server cluster:

Approximate round trip times in milli-seconds:
Minimum = 15ms, Maximum = 15ms, Average = 15ms

krbonne 02-03-2007 03:30 PM

European Server
 
Gegroet,


Quote:

Originally Posted by Tony (Post 5164)
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- 02-03-2007 05:59 PM

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

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


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

martin 02-03-2007 08:19 PM

Quote:

Originally Posted by krbonne (Post 5192)
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.


Quote:

Originally Posted by krbonne (Post 5192)
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.

Quote:

Originally Posted by krbonne (Post 5192)
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 02-03-2007 09:31 PM

Gegroet,


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



Quote:

Originally Posted by martin (Post 5199)
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 02-03-2007 09:52 PM

Quote:

Originally Posted by krbonne (Post 5201)
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 02-04-2007 12:26 PM

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 02-04-2007 02:18 PM

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 02-04-2007 09:34 PM

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 02-04-2007 10:39 PM

Quote:

Originally Posted by martin (Post 5228)
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 02-05-2007 02:33 PM

Quote:

Originally Posted by piscicelli (Post 5218)
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.

ctylor 02-06-2007 01:47 AM

Quote:

Originally Posted by martin (Post 5246)
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.

I am interested in making sure that Voxalot is not ever acting as an outbound proxy in case it detects my device cannot bypass Voxalot's NAT handling. I was curious if disabling "Enable symmetric NAT handling" (i.e. setting it to No) in my members details would have the effect of bypassing Voxalot acting as a middle-man in all situations it detects it is needed, and the calls either work or bust? I'd like to test to see if my STUN configuration is optimized or not.

martin 02-06-2007 02:06 AM

Quote:

Originally Posted by ctylor (Post 5269)
I am interested in making sure that Voxalot is not ever acting as an outbound proxy in case it detects my device cannot bypass Voxalot's NAT handling. I was curious if disabling "Enable symmetric NAT handling" (i.e. setting it to No) in my members details would have the effect of bypassing Voxalot acting as a middle-man in all situations it detects it is needed, and the calls either work or bust? I'd like to test to see if my STUN configuration is optimized or not.

Yes, setting this flag to "No" totally bypasses the NAT handling logic.

sim2sim 02-06-2007 06:55 AM

Hello, Martin.
Problem:
At connection voxalot.com dial plan works correctly.
At connection eu.voxalot.com it is not transferred Prefix, set in dial plan.
My login: 550555
Thanks.

martin 02-06-2007 07:07 AM

Interesting, the code is identical on all servers. Could you please send me the details of what it is you are testing.

Thanks.

sim2sim 02-06-2007 07:23 AM

Login: 233332
The fragment dial plan:
100 _ZXXXXXX 8495${EXTEN} SipNet Yes [Edit] [Delete]
Is typed 1000000
At connection voxalot.com (us.voxalot.com) passes a set 84951000000@sipnet.ru (a voice server of Moscow time).
At connection eu.voxalot.com passes a set 1000000@sipnet.ru (the subscriber does not exist).
P.S. You can test service under this login. Calls on all numbers of Moscow (8495, 8499) is free.

martin 02-07-2007 12:48 AM

Quote:

Originally Posted by sim2sim (Post 5275)
Login: 233332
The fragment dial plan:
100 _ZXXXXXX 8495${EXTEN} SipNet Yes [Edit] [Delete]
Is typed 1000000
At connection voxalot.com (us.voxalot.com) passes a set 84951000000@sipnet.ru (a voice server of Moscow time).
At connection eu.voxalot.com passes a set 1000000@sipnet.ru (the subscriber does not exist).
P.S. You can test service under this login. Calls on all numbers of Moscow (8495, 8499) is free.

I had a brief look and the logic should work the same way on all servers. That said, there is the possibility that the regular expression libraries differ.

I have raised an item in our problem tracking system to take a closer look and will advise.

martin 02-07-2007 06:39 AM

Quote:

Originally Posted by sim2sim (Post 5275)
Login: 233332
The fragment dial plan:
100 _ZXXXXXX 8495${EXTEN} SipNet Yes [Edit] [Delete]
Is typed 1000000
At connection voxalot.com (us.voxalot.com) passes a set 84951000000@sipnet.ru (a voice server of Moscow time).
At connection eu.voxalot.com passes a set 1000000@sipnet.ru (the subscriber does not exist).
P.S. You can test service under this login. Calls on all numbers of Moscow (8495, 8499) is free.

Seems we had a database replication problem with the EU server. Could you please retest and let us know if you still have a problem.

sim2sim 02-07-2007 07:13 AM

Now all works perfectly.
Thanks.

martin 02-07-2007 07:15 AM

Ok thanks sim2sim


All times are GMT. The time now is 11:31 AM.

Powered by vBulletin® Version 3.7.2
Copyright ©2000 - 2022, Jelsoft Enterprises Ltd.