Closed GoogleCodeExporter closed 9 years ago
You can try to activate STUN and ICE (it may solve media network routing
problems). (See the FAQ wiki page to see how to do that)
However could also be something with O2 that simply block VoIP SIP traffic.
I don't know the capabilities of this network, nor the things they allow you to
do (technically and legally), but carriers in France for example just block
this kind of traffic (technical blocking + legally prohibited).
If they block, unfortunately, nothing can be done on the application side.
Original comment by r3gis...@gmail.com
on 24 Nov 2010 at 10:14
I tried to turn on stun and ice. with ICE on, I don't notice any difference.
When i turn on STUN, I don't even hear the phone ring when I dial a number
anymore. Before i could hear it ring when I called someone, but could not hear
them when they answered. Now I hear neither.
Note that I've turned on STUN, but done nothing to the STUN server setting
since I understand that if you leave it blank, it has a default.
Second question: When I use it over wifi, the audio is very choppy. Can that be
improved?
Original comment by reganand...@gmail.com
on 25 Nov 2010 at 12:13
For stun, the default should be the counterpath stun server, if it's blank,
stun does nothing. (So you can try with counterpath stun server or ekiga's
one... or better if your sip provider give a stun server, you should use it.)
ICE without STUN activated does nothing interesting. So you should try with
STUN (with a configured server) + ICE.
For choppy sound on wifi :
Can be due to two things :
1- the HTC PSP bug (when screen goes off, wifi packets are not received in real
time), a workaround is to activate Keep awake while in call in settings > user
interface (before turn on expert settings mode in settings > (press menu key)
Expert mode.
2- low bandwith and use of a codec that consume a lot of bandwith. You should
try to allow only GSM codec for example (in settings > media > codec ,long
click to deactivate other codecs). Note that your sip provider must support his
codec else call will simply fail. (A good codec is iLBC, low bandwidth and good
quality, but sip providers doesn't support it in general).
Maybe it can also help on 3G to use a low bandwidth codec such as GSM codec.
Original comment by r3gis...@gmail.com
on 25 Nov 2010 at 12:22
Original comment by r3gis...@gmail.com
on 7 May 2011 at 12:22
Original issue reported on code.google.com by
reganand...@gmail.com
on 24 Nov 2010 at 10:08