I wondered about CPUs in terms of codecs. the CPU's in the phones don't seem to have enough mojo. That's my concern with investing $12 in G.729 for Zoiper, that I'll find the call quality still isn't good enough. In my experience though, Android+softphone combinations seemed to have a lot of latency regardless of device/softphone/codec combination used. I remember that from years ago, I'll give it another look. Is there one that works with voip.ms? quote: Yesterday I couldn't find a 3CX except for their own service. Thanks, I think I tried CSS a year ago but it was beta and I didn't find it as user friendly. My PBX happily negotiates an RTP port every time it sends an invite to my phone, and 9 times out of 10 there is bi-directional audio on the call.ĬSipSimple, Zoiper (IAX Support), 3CX, Bria. Some of your freezing, unreliability issues may stem from carrier's typical crippling of native Android functionality.Īlso contrary to popular belief or perhaps my experience was different if you are behind NAT there is no need to open inbound RTP ports within the range of the open RTP ports in your PBX as long as you configure your SIP settings in the PBX to allow NAT'ing for devices outside the local network from where the PBX resides. I never tested this on a stock Android ROM I've only done this with stock Cyanogenmod ROMs and corresponding radio/bootloader firmware. So far I've tested this on an HTC Sensation 4G running ICS and Jellybean, LG Optimus T running Gingerbread, and HTC Desire S also running Gingerbread. I have yet to miss a phone call to my Android phone via SIP ever since I put those two elements in place. It requires manually waking up the phone to resume connectivity. and by not qualifying the extension tied to my Android phone, so that the PBX isn't actively monitoring the connection status of the extension in the backend.Īs for the Cisco command, this tells the router to remove the UDP translation from the NAT table after 10 seconds of inactivity.įor some reason after a period of time since last SIP peer connection, the invites stop coming in. I solved it with one simple command in my Cisco router: Who would imagine that Android OS, made to run on phone, doesn't provide a way to wake up a phone application. in v2.2 Android there is no API to wake up registered SIP phone application when SIP request arrives via the network. And I guess it's not its fault, but rather it's a result of overall Android design (or rather lack of good one). Later phone may wake up and start ringing, but what's the point to ring in a couple of minutes of actual call, right? In some versions of Android it may run, in others CSipSimple may just suddenly crash time to time (as in my case). There are other (but free) good codecs, that support reduced bandwidth well, including GSM, SILK, ILBC, etc.ĭepending on hardware, Android OS and version of its kernel, it may work or it may refuse to get calls at the time they arrive because the phone is in sleep state. If your phone doesn't support it - no big deal. If it's not, you may still use CSS as a good test phone - it's very flexible in its configuration. You may try CSS with your particular phone and see if it's 'hit' in your case. Depending on hardware, Android OS and version of its kernel, it may work or it may refuse to get calls at the time they arrive because the phone is in sleep state. From my experience, it could be a hit or a miss in terms of reliability. I used to suggest CSipSimple here, but not now.
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