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Voxalot Support Support for the Voxalot service. |
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04-03-2010, 05:54 PM | #11 |
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Join Date: Dec 2009
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In sip setting I changed "Transport type" from "UDP" to "Auto" and "TCP"
After connecting with subscriber conversation duration was 32 sec and disconnected in both case. On duration of conversation I and my subscriber hear each other very well.... What have to I chenge in SIP setting for solve this problem? |
04-05-2010, 03:32 AM | #12 |
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Join Date: May 2009
Location: Canada
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My Nokia E51 works perfectly with Voxalot (same S60 FP1 phone like E71), I will post the settings tomorrow, it is late here now. Fring may be easier to configure, but it adds latency since all traffic is routed through Fring servers. It is an option if you have a phone without a native SIP client, but it so match for Nokia SIP. I will be back.
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04-05-2010, 12:54 PM | #13 |
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Join Date: May 2009
Location: Canada
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OK, here are the settings you need.
1. Go to Settings>Connection>SIP settings>New SIP profile Profile name> Voxalot (or whatever you want) Service profile > IETF Default access point > whatever you want (selecting 3G data might be problematic due to latency. WiFi is recommended) Public user name > sip:123456@us.voxalot.com (or eu.voxalot.com or au.us.voxalot.com); 123456=voxalot username, put in your username Use compression > No Registration > When needed (selecting Always ON will drain you battery quickly, not recommended if you don’t need SIP all the time) Use security > No Proxy Server Proxy Server address> sip:us.voxalot.com (or sip:eu / sip:au) Realm > voxalot.com User name > 123456 (your user name) Password > your password Allow loose routing > yes Transport type > Auto Port > 5060 Registrar Server Registrar Server address> sip:us.voxalot.com (or sip:eu / sip:au) Realm > voxalot.com User name > 123456 (your user name) Password > your password Transport type > Auto Port > 5060 2. Go to Nokia SIP VoIP Settings that you downloaded and sinstalled - VoIP services > New > use existing profile > select Voxalot, then enter profile Voxalot The only things you need to modify here are: Voxalot > Profile settings > Allow VoIP over WCDMA > ON if you want to call over 3G data Voxalot > Codecs > DELETE ALL profiles except G729. This doesn’t actually delete the codecs, it just sets G729 the default codec for Voxalot, the rest of the codecs are still present and will be used in case G729 fails. I use G729 for all my providers and I have no problems so far. If you want to call over 3G data, then G729 (and this setting) is mandatory, G711 is a no go over 3G/WCDMA. - NAT firewall settings > Domain parameters > voxalot.com > UDP NAT bind. Refresh 20 and CRLF refresh > ON With these settings I’ve had no problems whatsoever, my Nokia SIP client was rock solid and the reason I never needed a standalone ATA. Last edited by Corbu'; 04-05-2010 at 01:07 PM. |
04-07-2010, 08:10 PM | #14 |
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Join Date: Dec 2009
Posts: 11
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Corebu, thank....
This weekend I am going to solve problems with native SIP client - I don't have free times in work days.I noticed that appeared latency when I called through Fring SIP client - when I called through native SIP the quality of sound don't distinguish than quality of mobile or landline's phone |
04-10-2010, 03:58 PM | #15 |
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Join Date: Dec 2009
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I tried to use your setting - nothing helped - after 32 seconds from beginning connection subscriber disconnected......But when I use native SIP through VOIPRAIDER - all works very well...Which "bad of nail" with voxalot and native Nokia SIP
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04-11-2010, 12:56 AM | #16 |
Member
Join Date: May 2009
Location: Canada
Posts: 81
Thanks: 14 Thanked 6 Times in 5 Posts |
It shouldn't disconnect, maybe you have a problem with your router. Try at a different location with a different router or over 3G data.
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04-11-2010, 09:40 AM | #17 |
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Join Date: Dec 2009
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I don't have router....I use Megafon (my operators in Russai) 3G unlimited Internet plane wint my E71 - I don't know about structure network of my operator and I don't know about network setting of my mobile phone
When I set up SIP native client for VoipRaider or WebDirectCall my subscriber certain doesn't see my CallierID but call session doesn't disconnect after 32 ceconds...... I don't believe that problem in router - other two VOIP operators don't disconnect after 32 sec and work very well...... |
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