I'm able to bypass callcentric by moving the GSM codec (pre-cursor of the AMR codec) on top of iLBC in the CSS codec priority list. This will lead to the GSM codec to be picked for incoming calls, but iLBC for outgoing calls. GSM does not sound as good as iLBC, but it is still very useable. Interestingly, iLBC seems to work for incoming calls when using the Media5-fone SIP client (again, without iLBC in the middle).

Asterisk SVN-trunk-r122766 built by root @ usflhi01a.foo.com on a i686 running Linux on 2008-06-14 23:12:08 UTC usflhi01a*CLI> I also ran the get_ilbc_source.sh script and re-compiled asterisk as well as re-started it.

I did a little bit of digging and it appears that the problem may actually lie with CSS/pjsip rather than PBXes. A wireshark trace shows that the MIME types in the SDP exchange between CSS and PBXes differ, i.e. 'iLBC' for PBXes (which follows the IETF recommendation) vs 'ILBC' for CSS. Depending on how CSS/pjsip handles the SDP messages it could very well reject the iLBC codec depending on MIME type mismatch. I could also be a iLBC mode negotiation problem (20ms vs 30ms ptime). A bug report may in in order (just as with G.726 as mentioned in an earlier post of mine).

Quote: Originally Posted by chdloc Interestingly, iLBC seems to work for incoming calls when using the Media5-fone SIP client (again, without iLBC in the middle). I did a little bit of digging and it appears that the problem may actually lie with CSS/pjsip rather than PBXes.

A wireshark trace shows that the MIME types in the SDP exchange between CSS and PBXes differ, i.e. 'iLBC' for PBXes (which follows the IETF recommendation) vs 'ILBC' for CSS. Depending on how CSS/pjsip handles the SDP messages it could very well reject the iLBC codec depending on MIME type mismatch.

I could also be a iLBC mode negotiation problem (20ms vs 30ms ptime). A bug report may in in order (just as with G.726 as mentioned in an earlier post of mine). Just thought I'd share my setup which has been very good so far.

I use the above server with the built in sip client on android. What I like about this setup is true stock dialer integration.

Real hardware proximity sensor support and super simple setup. Just put in your Google voice info (I'd suggest 2 factor authentication token) and just add a new account with generated user/password/server into internet calling option in android's dialer. Dead rising 2 cheats 360.

It uses UDP and PCMU codec. Not the greatest but it may work well for most. I know I didn't see much difference between codecs.

Works for incoming/outgoing and there is no software to install and nothing more to configure. I still want to thank op for posting the info. It was certainly educational. BTW, consider donating for this guy if you're going to use the server if you're still using it after a while. Also, main reason to use this over pbxes.org is that pbxes.org free account has a 30 minute re-try on network failure and simonics re-tries constantly until it gets a hold of your client. Sent from my Nexus 4. Most of those settings are good, but I've spoken to the CSipSimple developer personally and raising the Clock Rate to anything higher than what the codec offers is counterproductive.

If you're using a 8kHz codec -- PCMU which is also known as G711.u, G729, Speex, GSM, etc. Then make sure the Clock Rate is set to 8kHz. Increasing that setting will actually degrade the audio quality. If you're using G722, which is 16kHz, then change it to 16kHz. Also, the MAIN thing for quality is to run an echo test and adjust the mic gain and earpiece volume while performing this echo test. You can access these settings in call by clicking the wrench in the lower right corner.