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Posts: 306 | Thanked: 106 times | Joined on Feb 2010
#21
Guys,

I have been in touch with Olivier Crete (from Collabora) regarding this bug via irc. He looks after the voice quality issues in N900. I setup a freeswitch server against which Olivier did some siptraces.

He believes that there is a bug in gstreamer rtp jitter buffer (gstrtpjitterbuffer) library in N900. That bit of the code is very touchy. He is the person responsible to fix the code in N900 but presently he has no free time. I did ask him when he could fix this bug but he did not give any time-frame.

I did carry out tests using empathy (on PC) which also uses the gstreamer and telepathy-sofiasip library (same as N900) but the voice was very clear. It would seem that the implementation on N900 has issues. Apparently, the same upstream code will be used in meego as well, so nothing exciting on that front.

I also asked Nokia Wifi developer for recommendations on any good router (with sip calls) which support Power Saving Mode (PSM) but the response wasnt very useful
Code:
We have bunch of APs from different vendors: Cisco, Linksys, Dlink, Zyxel. I don't think they have been tested specifically for sip calls.

Some models have some problems. Unfortunately, there is no good list for those since different versions have different problems.

In general newer models perform better.

Br, Janne
Hopefully, Olivier will be able to fix this bug some time in the future. If anybody understands gstreamer please have a look at gstrtpjitterbuffer.
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Voice choppy on sip calls
Please vote for bug number 10388
 

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#22
Originally Posted by alex25 View Post
I just found out that the sound is much less choppy using sip if the phone is overclocked to 900MHz. So it seems to be definitely a performance issue.
I have this choppy sound for WLAN and 3G sip calls. I use the g729 codec.
g729 is VERY cpu-intensive.
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"Impossible is not in the Maemo vocabulary" - Caballero
 
Posts: 7 | Thanked: 5 times | Joined on Aug 2010
#23
Hello all,

Where do I change the codec on my phone?

Does it need changed on the server (asterisk) side?

-Eric
 
Posts: 59 | Thanked: 36 times | Joined on Nov 2009
#24
Originally Posted by ewheelerinc View Post
Hello all,

Where do I change the codec on my phone?

Does it need changed on the server (asterisk) side?

-Eric
Best is to change it on the server side. E.g. for using ilbc add the following to your asterisk settings:
Code:
disallow=all
allow=ilbc
 

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Posts: 59 | Thanked: 36 times | Joined on Nov 2009
#25
I the meantime I found a solution that works for me. I use the ilbc codec for a few weeks now without choppy audio.
To archive this I made the following two changes to my n900:
1.) Install the new pulseaudio package provided by the user MohammadAG as discussed here
2.) Set the processor speed to 600MHz (fix) during sip calls (I think the stock kernel does this, however it seems not to work with the power Kernel).
To set the processor speed to 600MHz during a sip call I use a dbus script. I added the following config file to the /etc/dbus-scripts.d/ directory with the following content:
Code:
/usr/sbin/speedadaptnormal * * com.nokia.mce.signal sig_call_state_ind active *
/usr/sbin/speedadaptfix * * com.nokia.mce.signal sig_call_state_ind none *
And created the two files in /usr/sbin/.
speedadaptfix
Code:
#!/bin/bash
echo 125000 250000 500000 550000 > /sys/devices/system/cpu/cpu0/cpufreq/ondemand/avoid_frequencies
kernel-config limits 600 600
speedadaptnormal
Code:
#!/bin/bash
echo > /sys/devices/system/cpu/cpu0/cpufreq/ondemand/avoid_frequencies
kernel-config limits 125 1000
As said before this works perfect for me. No choppy audio anymore!
 

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Posts: 306 | Thanked: 106 times | Joined on Feb 2010
#26
Ok guys I have a disappointing news from Collabora dev. This bug is not fixed yet and so will miss the 1.3 firmware release!
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Voice choppy on sip calls
Please vote for bug number 10388
 
Posts: 3 | Thanked: 1 time | Joined on Sep 2010
#27
I just installed the libpulse package provided by brad112358 in the thread alex25 linked. No other changes, stock kernel.
And for the first time since I owned the N900, I can actually use it with Skype!
Only tested Skype-to-Skype over WLAN, but it worked for the ten minutes I tested.
Thanks for the link, alex25!
 
Posts: 2 | Thanked: 0 times | Joined on Oct 2010
#28
I also installed brad112358's libpulse0 package (linked at http://talk.maemo.org/showpost.php?p...&postcount=113) that was referenced on the 12th page of the thread that alex25 referred to. Stuttering is as bad as ever.

The only difference I can see is that I'm running the power kernel (for kismet). (Well, and that despite many hours troubleshooting this problem, it's still unusable it with Skype/VoIP)

I also tried MohammadAG's multipackage approach that is the topic of the thread that alex25 directs to. Stuttering did not improve, and I ran into the other expected problems (ringtones, etc.)

Can anyone here comment on whether using a different program would bypass gstrtpjitterbuffer? I noticed that SFLPhone has a mispackaged, non-version-matching n900 port, but it may be more fruitful to try to straighten that out than applying ineffective patch after patch.

Last edited by acabre; 2010-10-04 at 15:10. Reason: removed emoticons
 
Posts: 59 | Thanked: 36 times | Joined on Nov 2009
#29
Originally Posted by acabre View Post
I also installed brad112358's libpulse0 package (linked at http://talk.maemo.org/showpost.php?p...&postcount=113) that was referenced on the 12th page of the thread that alex25 referred to. Stuttering is as bad as ever.

The only difference I can see is that I'm running the power kernel (for kismet). (Well, and that despite many hours troubleshooting this problem, it's still unusable it with Skype/VoIP)

I also tried MohammadAG's multipackage approach that is the topic of the thread that alex25 directs to. Stuttering did not improve, and I ran into the other expected problems (ringtones, etc.)

Can anyone here comment on whether using a different program would bypass gstrtpjitterbuffer? I noticed that SFLPhone has a mispackaged, non-version-matching n900 port, but it may be more fruitful to try to straighten that out than applying ineffective patch after patch.
Did you set the CPU clock rate to a fix value during the SIP call?
(execute this command during an active call: kernel-config limits 600 600)

It works perfect without choppy audio on my setup.
 
Posts: 187 | Thanked: 41 times | Joined on May 2010
#30
it works fine over 3g , using wireless the audio is really choppy
 
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