#1453 Bad audio, WPA issue ?

[Anonymous]

Originally created by: cmendio...@gmail.com

What steps will reproduce the problem?
1. Install 0.03-1 on MB860/Froyo
2. Call, bad audio ...
3. Install trunk 1158
4. WPA2-PSK access no longer works

What is the expected output? What do you see instead?

What version of the product are you using? On what operating system?

Please provide any additional information below.
I installed 0.03-01 on a MB860 running Froyo, and it registered right away but I could not get audio to behave. G711 (and G729 too) sounded
like the clock was too fast on the phone: slow/grave on the other side
and squirrilish on the mobile.

So I tried trunk (1158) and it somehow got in the way of my WPA2 PSK access. :(
I can connect to an open AP, but fails on WPA2 PSK. Any idea ?
I'm new to android in general, but no so on linux, voip and networks.

[Anonymous]

Originally posted by: r3gis...@gmail.com

Mmm, the point about the clock rate of audio sounds to me something with the android audio layer. You should try to play with audio troubleshooting options see: https://code.google.com/p/csipsimple/wiki/FAQ?wl=en#Audio_routing_troubleshooting

About the access point problem that's weird. It could be linked to UDP packet size that are too big (if your sip provider does support that, I'd advise to swicht to compressed mode -- it's in network settings).
If it does not helps, could you try to collect some logs (see HowToCollectLogs wiki page). Maybe there is some interesting clue about what's going wrong.

Status: Need-Details

[Anonymous]

Originally posted by: cmendio...@gmail.com

Ok, I guess I have some more info, but still lacking.
On the connection info, problem *seems* to be some issue with the Atrix when reconnecting fast with WPA2. Like it tries to reuse some old key. It might also be the other side (AP=OpenWRT Kamikaze). If I keep wi-fi off for some time, then it connects ok. But I found this issue only after installing 1158, thus my previous post.
Might it be that csipsimple alters in some way when the mobile connects or disconnects?

Audio: now that I manage to get connected again, 1158 default settings do not show
the fast/slow of 0.03-01, but a long buffering both ways, some 500ms ?
Disabled Focus Audio and it improved a lot. WebRTC sounded weird.
Is there a document that says what do those options do ? (Other than code :)

Thanks.

[Anonymous]

Originally posted by: r3gis...@gmail.com

You can read my comment here :
https://code.google.com/p/csipsimple/issues/detail?id=119#c38
(comment 38)
It explains the current state of my empiric results after testing a lot of devices and getting a lot of feedback from users.

About the motorola atrix I had known issues in the first version of Motorola ROM (motorola engineers did a fix after the release about that).
So there is specific conf for the atrix, depending on the ROM you are running, you may have a look here and try proposed options :
https://code.google.com/p/csipsimple/issues/detail?id=795&q=atrix
The main idea is to use the micro source "CAMCORDER" (depending on the ROM) and a clock rate of 32kHz.

I do not put down a doc about that somewhere for the simple reason that this should not behaves like that. That's just workarounds to try to get things working with weird things done by manufacturers. The doc of each option is actually in android SDK audio manager section. However, between what's in android sdk doc and what you observe on each phone there is sometimes a lot of difference. So consequences of activating each option has various result depending on the device and the ROM. For example I know that samsung has almost each of its device considering the audio api a different way (I guess they have different team for audio drivers). Motorola did weird things on the atrix (I guess due to the way it interface with their dock), HTC has globally something that match apis (probably because they worked a lot with google on the beginning of android), but their upgrades of 2.3 have some constant problem which is easy to solve. Usually Nexus devices from google behaves very good regarding audio api. The latency on the galaxy nexus is very very low.

The latency really depends on the ROM and the manufacturer. There is some new options from 2.3 that should ensure lower latency but few devices actually has that implemented. And sometimes using these options (the in_communication options) result in worse audio. In fact, before android 2.3 nothing was really done in android to do actual voice over IP application. It's only from 2.3 when google had to introduced their stock sip application that they started to think about improving the audio stack. But they also are aware that manufacturer do crappy things. If you have a look to the google webRTC project you'll see a lot of if/else just to workaround samsung devices.

So to try to deal with this fragmentation of the audio layer, the application try to automatically setup best settings for the device based on infos it can get about the device it's running on.
So if you find something interesting, some better options for your device, do not hesitate to share (send me logs so that I can get the exact device name) and I'll add something in the app to autoconfigure audio layer for this kind of device so that other users with your device will have it working out of the box ;)

[Anonymous]

Originally posted by: cmendio...@gmail.com

Sweet. Thank you. I've been reading some related info, and following links.
It seems Android is far from making it easy for voip :-/
So I don't know why but "Focus Audio" will add to latency. That one I don't remember being documented anywhere.
Also, my first test of speakerphone yielded high white noise/static on the other side. I was not able to reproduce it, it solved by disabling/enabling speaker.
I'm on 2.2.2, not shure yet how to jump into 2.3 or 4 ? :) But that's unrelated.