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New User Questions

General Discussion about AGEphone Mobile that doesn't belong in the categories below.

New User Questions

Postby mgerbasio on Mon Mar 30, 2009 6:25 am

Hi,
I just downloaded your software v2.75 on an AT&T Tilt using a Gizmo account. I have a couple of questions and couldn't find an answer through searching.

1. When I'm connected to my PC with a USB cable I have a data connection to the internet, but I can't connect to the SIP account. Not a big deal but since wifi kills the battery, I'd like to leave my phone plugged into my PC as much as possible.

2. When I speak, the person on the other end doesn't hear me.

3. The trial mode of one minute is too short to really evaluate the software. Something like a week of 10 minutes, then one minute trial would work better.

4. Is there a document explaining some settings to get better voice quality when on wifi vs 3G?

Thanks.

Regards-Michael G.
mgerbasio
 
Posts: 1
Joined: Mon Mar 30, 2009 5:41 am

Postby Michael on Thu Apr 02, 2009 11:04 pm

Hi Michael and welcome to the forums! You raise some good questions that I hope I can match with some good answers... Let's see!

1. Your mobile uses the ActiveSync connection which is fine for certain tasks like downloading mail and browsing the internet. Unfortunately, it is completely unusable for more complex protocols like SIP and thus can't be used to make VoIP calls.

2. This sounds like a network issue. AGEphone Mobile reports a "... NAT" or "Blocked" message on startup if left to the default options. If you see this message, please let me know what it is. If you do not get the message, please go to "Menu - Settings - Network" and change "NAT Traversal" to "STUN / UPnP". If it is already set, please change the STUN server to one listed in this wiki.

3. We are always open to suggestions in this matter and may offer different evaluation methods in the future.

4. There is no such document and instead I'll just list the few settings that influence the call quality here. For starters, you can try lowering the buffers under audio a bit which will take out much of the delay that many VoIP calls bring with them. If you want to go a bit farther, close AGEphone Mobile and open the "sipd.conf" under "My Documents / My Phone Booth". Under [General] you will find a string called "MediaTypes=3 0 8" where

3 = GSM-FR
0 = G.711u
8 = G.711a
101 = DTMF Setting (do not change!)

You can rearrange those numbers and if you take out the "3" AGEphone Mobile will always use G.711 (64 kbit/s) which - while using a higher bitrate - provides better audio quality than the GSM-FR (13 kbit/s) codec. Depending on your 3G connection you may or may not be able to use G.711 as you have to add 20 kbit/s overhead to the audio stream in both directions. WiFi (and the attached landline internet connections) should be able to provide the necessary upstream in nearly any case.
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Michael
Site Admin
 
Posts: 195
Joined: Thu Apr 03, 2008 6:12 pm
Location: Kyoto, Japan


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