My callers & receivers of my calls report lots of static on their end of the line

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What steps will reproduce the problem?
1. Register sipsorcery account in csipsimple
2. Start a call to another party / Receive call from another party
3. Start phone conversation

What is the expected output? What do you see instead?
I expect a smooth phone conversation between myself and other party on the 
call. The other party on the call reports lots of static / clicking/ shooting 
sound.

What version of the product are you using? On what operating system?
I'm using version 0.02-03 (current market version) on Android 2.3.7 
(CyanogenMod 7.1)

Please provide any additional information below.
I'm running CM7.1 (android 2.3.7) on HTC Nexus One running the current version 
of the app on the market.

I manage my I can use my sipgate account directly or use it via sipsorcery. I 
mostly use sipsorcery to take advantage of dialplans and also get access to the 
google voice callback api. So gv is my main number & I have a data only plan. I 
use the sipsorcery profile in the app with ICE and STUN turned on. My friend 
always complains about a static / clicking / shooting sound every time we're 
talking. this happen with both my data connection on the phone & any wifi 
connection I happen to connect my phone to. On the PC, I use Blink to setup & 
use sip/voip but my friend does not complain about static.

Is there anything you can recommend to help me fix this - if any. It is a known 
bug with a possible fix?

Thanks for your hard work & dedication to building the best sip/voip client on 
android. it makes the built in gingerbread client look and function like a 
stone age tool. 

Original issue reported on code.google.com by oky...@gmail.com on 11 Oct 2011 at 6:09

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I had this happen and finally I activated the lowest bandwidth G729 codec and 
it is MUCH better. Apparently the phone cannot handle the higher quality codecs 
at all.

Original comment by tommy.tr...@gmail.com on 14 Oct 2011 at 5:36

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My VoIP provider (VoIP.ms) also accepts GSM so I just chose GSM 8khz and it 
seems to work well, too.

Original comment by tommy.tr...@gmail.com on 14 Oct 2011 at 5:50

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Or it can also be a matter of bandwidth available.

If you make a call over networks with low bandwidth using PCMU/PCMA codec it 
will necessarily procude very bad results. PCMU/PCMA is 64kbps codec (in fact 
uncompressed). There is no loss in audio, it's 8kHz (so not the best quality). 
On the other side it requires less CPU processing since no compression. 
There is other codecs optimized for bandwidth usage. G729, indeed (8kbps), iLBC 
(13kbps), GSM (13kbps) a lot of loss.
There is other codecs that allow higher clock rate (and higher quality), speex 
16 and 32kHz, g722 ...

So it really matter depends on your network and its capabilities. It also 
depends on what is supported on the other side. Unfortunately, sip providers 
often only supports pcmu/pcma (aka g711). 
You can play with codec priorities in the two list (low bandwidth and high 
bandwidth) in order to adjust which codec should be prefered. However, it 
should be done with care cause the application does not always have the last 
world on the choice, so sometimes worth to disable all but one you'd like to 
force.

The choice of the codec is always complex. Bandwidth/Quality/CPU usage. (On 
nexus one CPU usage should not matter so much).

Also, good to know as talking about codecs... CSipSimple nightly build now 
support AMR codec (for android 2.3 and upper) which can run at very low bitrate.

Original comment by r3gis...@gmail.com on 14 Oct 2011 at 5:54

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Ok guys, thanks for all your feedback. I'm new to sip/voip so the whole talk 
about codec and clock rate as applied to this subject sounds foreign to me. So 
help me clear something here: you guys think I should select a clock rate 
first, then select a codec that matches the clock rate I chose before? Can a 
codec that clocks at a lower rate work with a higher chosen clock rate and 
vice-versa? 

If any, please direct me to a manual or website where I can read more about 
codecs and clock rates; and how to set them to get the best results. I really 
want to educate myself on this as this is the only way I make & receive calls 
on my data only plan. Thanks in advance.

Original comment by oky...@gmail.com on 14 Oct 2011 at 7:57

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Yes sorry I was not very clear this clock rate stuff.

The first thing to know is that codec clock rate and android audio clock rate 
are independant.
Opening the android audio clock rate at higher rate produce sound of better 
quality (at least percepted as better), even if codec only support lower clock 
rate.
The application adapt the codec clock rate to the audio driver clock rate.

For audio driver clock rate there is a setting in csipsimple about that. It's 
set to 16kHz by default on high CPU devices, and to 8kHz on lower. On device 
with low CPU (it is not your case), it is better to have the same clock rate 
for android audio layer than the one used for the codec. It avoids adaptation 
which consume CPU.

So in your case, you can keep 16kHz, it should be fine for all codecs.

Then, there is the codec. The codec is used to send your audio packet through 
the network. PCMU/PCMA does not compress packets and work at 8kHz. It results 
that it use 64kbps of bandwidth, which is important and not very suitable for 
mobile connections.
There is other codecs. All others do compression on voice packets. Sometimes 
compression make quality lower. For example GSM codec (the one that is also 
used by your phone for GSM calls), is known to be very lossy and produce worse 
quality. However, if your bandwidth is low, you can get better results with a 
codec that compress a lot and reduce quality than with a codec that does 
compress nothing but which packet are not transmitted in time through the 
network. Usually in this case, what is observed is what you describe : static, 
cuts & choppy.
There is also codecs -such as g722- that compress packet but work at higher 
clock rate. It result higher bandwidth usage, but voice quality is better (if 
bandwidth is big enough, else it will just be more choppy).

So, what you should do concretely :
1st : check with your SIP provider which codec they support.
2nd : according to this list, choose the one that fit the best your network 
bandwidth. (You can give here the list of codecs supported by your provider, 
I'll give you advise on the best). Obviously if they only support PCMU/A (aka 
G711), there nothing to improve.
3rd : setup csipsimple to prefer this codec among others, and if necessary 
force the selected codec to be the only one available.

Original comment by r3gis...@gmail.com on 15 Oct 2011 at 6:51

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Hey Regis, 

Thanks for all the education you gave me in your last post/comment. Things make 
much more sense to me now. 
I contacted sipgate in regards to the recommended and supported codecs. This is 
what they wrote back:

"G729 is recommended, if your phone does not support G729, try another codec. A
list of codecs supported by sipgate follows here:

G.729

G.711

iLBC

GSM

G726

Please feel free to try each codec in turn and see which one works best for your
connection."

1. Based on this respond, what do you recommend that I use when 
(a) I'm using my mobile data - T-Mobile's HSPA network (USA)
(b) I'm on my home's wifi. Our router is also connected to a Vonage adapter for 
our home phone.

2. Is there a way to test to know how much bandwidth my phone is getting in 
either case? I have no idea how much bandwidth my phone gets in either case 
that's why I ask.

3. Should I turn on QoS in my router settings? I read somewhere that it can 
help improve the quality of voip calls on my home network.

4. I plan to upgrade to a faster phone with a faster processor, more memory and 
a faster data transfer (T-mobile's theoretical 42 mbps '4G'). It will either be 
the Amaze 4G from T-mobile or Samsung Nexus (Nexus Prime) if it comes to 
T-Mobile USA and the specs are as good as rumored. My question is, will there 
be a significant boost in my setup & sip calls (using csipsimple) with a better 
spec'd phone?

Thanks for your time developing this kick ass app and your dedication in 
supporting it. I salute you.

Original comment by oky...@gmail.com on 17 Oct 2011 at 4:35

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Ok,

g729 is supported by csipsimple, but it requires a license use for end users 
(that I do not provide, since this is just an opensource application). However, 
if you get the license by your own (or if you already have another app that 
grant this license use on another app you can activate it in codec list). 

If you don't want to use this codec cause of its license terms (which I advise, 
because I think that it is better to use an opensource codec than something 
proprietary, but that's just my point of view). 

You can try iLBC as first choice. It is almost equivalent to g729 (quality and 
bandwidth). It should be suitable for both wifi and mobile connection.
The only problem is that all sip providers does not support it very well (cause 
it requires many CPU). But if your provider has it correctly supported it 
should be fine.
If it does not work you can try GSM codec. Voice quality will not be crystal 
clear, but should be good enough. This codec is the same used on standard GSM 
mobile calls, so it is not so bad ;).

g726 is not supported by csipsimple for now, but its not really suitable for 
mobile anyway. And for g711 (aka PCMU/PCMA), I recommand to disable it on 
mobile data and depending on wifi connectivity can be suitable for wifi.

About the bandwidth test, you can download for example : 
https://market.android.com/details?id=com.ookla.fccbroadband&hl=fr (that's just 
the first hit I found while searching, not sure it is the best). The point is 
that there is apps that allow to test the connectivity.

About QoS, it depends on network equipments between you and the sip server. It 
does not always improve things (sometimes it make things going worse). But it 
is indeed interesting to try. In CSipSImple this setting is in expert settings. 
So to get the option => Go in settings, press menu key of the phone and select 
"Expert mode". Then go in Settings > Network > Check enable QoS.

And for the last point...well, compared to a nexus one, CPU speed increase 
should not be significant for voice over IP. The CPU of the nexus one is 
already really fair for voice over ip. The point that could help is 
connectivity. I never tried with 4G, but if real bandwidth is as good as the 
theoretical one, should indeed improve things.

Original comment by r3gis...@gmail.com on 18 Oct 2011 at 11:29

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Thanks very much for your help on this issue. I'm going with the iLBC and GSM 
in that order. People that I talk to do not complain about the choppy/static 
sound anymore. I do not plan to purchase a license for G.729 but I was 
wondering if I already have the license by purchasing the nexus one or an 
application on mobile (GrooveIP, maybe?) How can I tell?

Lastly, what does the "Voice Audio Detection" setting do?

Again, thanks for this wonderful application. Some donation will be coming your 
way soon. 

Original comment by oky...@gmail.com on 18 Oct 2011 at 3:55

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Donation sent.

Original comment by oky...@gmail.com on 18 Oct 2011 at 4:31

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Thx for the donation :).

Voice auto detection feature detects when you do not talk. When it detects that 
there is no voice, it does not send packets at all. It allow to reduce 
bandwidth used.

Original comment by r3gis...@gmail.com on 7 Nov 2011 at 12:23

• Changed state: Fixed