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I'd also like to know how to do this, and I'm shocked that the speex codec is not supported by the native dialer. g711.u (Ulaw) is the default codec, and is simply not usable while driving. GSM works well over 3g/4g on my GSM Galaxy Nexus over T-Mobile, but it sounds like you're talking through a tin can. g711.u is great over WiFI, but WiFI is not always available. Using the integrated SIP phone is much better than using CSipSimple and sipDroid. When CSipSimple works, it's great. Speex sounds great over 3g, but CSipSimple is very tempermental.
 

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Thanks for the reply and advice, I'm actually running a Google Voice account through PBXes.org or I would try to roll my own server. Something that has actually helped the stock dialer to become more bearable is switching to UDP rather than TCP -- I guess UDP deals better with packet loss, so a semi-shoddy connection doesn't completely shut down a VOIP stream (whereas the same connection using TCP would get screwed over by any packet loss). Thanks again!
Sorry for bringing up such an old topic, but I figured I would help you out. I have made a lot of progress since my original post. With the right settings, CSipSimple is awesome, and actually sounds better than some hardware VoIP phones. If you decide to roll your own Asterisk server, I suggest you try using TCP transport for SIP signaling, the G729 codec for mobile data, and the G722 codec for WiFi. There is a huge difference between in battery consumption by using TCP over UDP, especially over WiFi. There is a ton of information here, and it covers how you can set up an Asterisk server with Google Voice, TCP transport, and the G729 codec. I've been using this since the end of September, and it's great.
 
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