SIPDroid is a java based, open source SIP client that has recently been developed for use with mobile devices based on Google’s Android platform.
Based upon a Java SIP stack contributed by MJSip, SIPDroid is currently in public beta.
The SIPDroid Users forum can be found here. The SIPDroid Developers forum is located here.
From the SIPDroid.org website:
After completion of the closed alpha stage this project will publish the software for free under the terms of GNU General Public License v3. The first beta version will be for software testing. So please allow for some issues and incompatibilities at the beginning.
Although SIPDroid will likely mature quickly, it is currently only fully supported using virtual PBX service from PBXes.com. PBXes.com offers a free basic account registration for their service.
Once you have created a basic account with PBXes.com you can set up additional SIP providers/registrars within the Trunks section of their web based UI.
PBXes allows you to register several trunks from multiple telephony service providers of your choice. PBXes routes incoming calls over SIP and the PSTN to you. If you are online you can take a call as VoIP, and if you are offline the call will fall back to GSM.
Beyond their free basic service, a paid account additionally allows for handoff of calls between networks. PBXes also supports NAT.
To install Sipdroid you need version 1.5 “Cupcake” of Android. It is already available from HTC for Android Developer Phones. Visit this link for details on updating the OS. An OTA (over the air) update for the other phones has been announced for coming in the beginning of May.
Tragically, most of us here at The VoIP Insider are Apple iPhone users, but we have procured an Android mobile phone from Yannick Tessier, our head of engineering, for testing purposes. We will attempt to get SIPDroid working this week with Asterisk and let you know how we fare.
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I have an ADP1 phone w/ 1.5 firware and have been trying to hook sipdroid to an asterisk server. I havent' quite gotten it yet. I am behind a NAT from the server, which seems to be part of the problem. I can a successfull registration and incoming calls ring and open, but no sound. Outgoing calls "abnormally terminate" at dial time.
I would love to hear from anybody who has gotten farther and any adjustments you've made.
Interesting that SIPdroid essentially requires the use of PBXes.com while the lead developer, Pascal Merle is listed as the CEO of PBXes.com.
https://www2.pbxes.com/iptel_impressum_e.html
Not really a huge deal but explains why the development appears to be focused on PBXes.com rather than Asterisk or any other standard SIP-based PBX.
Partial success, I managed to call Sipdroid ok... once.
ADP1 w/1.5, AT&T =>EDGE only, Asterisk on fixed IP (Speakeasy). Key settings in sip.conf: nat=yes insecure=invite,port qualify=no (Sipdroid sets the sender IP to 127.0.0.1, and doesn't seem to handle Asterisk's "pings").
Significant/perceptible latency, otherwise the sound quality was definitely acceptable. Clearly there is potential!
Not been able to place any call from Sipdroid so far however... :/
Sipdroid worked fine for placing a call after I read the pbxes.com Getting Started guide. I set up an extension and a trunk with sipphone, and it sounded pretty good!
Also, pbxes.com is a cool service - impressive to see telephony folks implementing the very flexible business models of mailbox and mailing list hosting companies. I hope that getting it working with other sip providers is mostly a matter of testing and tweaking.
Sipdroid work fine with my asterisk server:
Registration OK (without password at this time)
Place a call OK sound quality OK
Receive call Partialy OK : the phone display the incoming call , I can answer , but the phone don't ring
my sip.conf
[205]
callerid=
canreinvite=no
context=sip
dtmfmode=auto
host=dynamic
language=fr
nat=yes
port=5060
qualify=no
record_in=Never
record_out=Never
type=friend
username=205
where did you edit your sip.conf?
"the development appears to be focused on PBXes.com rather than Asterisk"
PBXes runs on asterisk so integrating it into Asterisk/Trix/Elastix should be a walk in the park
Hi.. i installed sipdroid in my android emulator.. i can perform calls but i cant receive calls. the phone also does not display incoming call. can you help me?
Flawlessly running using UDP protocol over Verizon 3G network connecting directly to Asterisk 1.4.2.
Will this sip client divert audio to bluetooth? Can it Allow bluetooth to make voip call?