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Adding SIP and VOIP stack?

I want to be able to do voip through skype and I know I can add the iskoot number to myfaves and get free incoming but can I use it for free outgoing?
If this isn't possible how hard would it be to bring the VOIP anr SIP stack from another phone into the G1?

There are already 2 sip stacks for Android. Someone would just need to make a usable application for them.
http://blog.roychowdhury.org/2008/03/10/we-have-sip-working-on-android/
http://jeanderuelle.blogspot.com/2008/10/jain-sip-is-working-on-top-of-android.html
I have not looked recently for user applications, but the sip stack is definitely already there.

I looked into those SIP stacks, and the first one listed has a User Agent in the code.
The problem is that the Android API has no way to stream sound from the mic, and can't seem to stream data from RAM to the audio output (so you can do, say, audio decompression of the RTP stream). Or at least, there isn't a straightforward apparent way to do it...
So until one can get audio into/out of memory (and not by writing to a file), this isn't going to be happening anytime soon, sadly. What may be interesting is to find out how these call recorders work...but again, I think that's all to a file.

What about StreamFurious? It streams shoutcast, so this must be possible in some way.

From network to speaker, it is possible...if you limit yourself to the codecs Android supports. However, VoIP uses some codecs that aren't supported by Opencore (at least, as best as I can tell)...so you'd have to code them yourself in Java (not hard for the PCM codecs, trickier for Speex and iLBC)...but then you need to play the uncompressed data from your app, and you can't seem to do that (unless you write it to a file and play that)
The recording side is worse, it needs to be recorded to a file. So making it a "phone" with the current API on Android is painful at best.

A SIP client working on Cupcake has been released today -- http://sipdroid.org

Integration within the OS and calling is working pretty good. The only downside is, that people can hear me, but I can't hear them :-/

I have no idea what I am doing. I spent 20 minutes messing around with sipdroid and I would guess about 19 of them were wasted. I will wait till this requires less setup (or its not 2am and I have a few extra brain cells to spare for figuring this out). Good to know its being worked on though.

I did test the SIP client, too.
First of all: thanks for this project. Together with sipgate.de and pbxes.org it works super well. I tested many calls and I can hear the other side and vice versa.
What did not work: entering directly sipgate.de in the SIP client (couldn't hear anything during a call in both directions).
What I hope will get better: better incoming call recognition (if device is in sleep mode the screen wouldn't turn on), and other small things (like to be able to turn off the client instead of turning off wireless options) and few other things which I don't remember now.
I am sure it will get better and better. So: THUMBS UP!

Was able to login to gizmo account but no audio. Needs option to turn sipdroid on/off, only way to do that right now is to turn off wifi or disable wifi in sipdroid.

Skype for WIndows Mobile

Has anyone had any luck with getting this to work well? Seems to always be like a delay on a call like when you call a country far away and you have to wait a few seconds to respond, but if I call via skype on a PC to the same skype user it's fine..
Also it seems by default the sound comes out of the back speaker rather than the ear piece, any way to change that?

I never tried to make any international call but it work's fine when I call to my home city (~300 km away from my location).
Regarding the sound,
this tool redirects the sound correctly where you want it, it can be found in the "Cabs repository"-thread but here's the link:
http://rapidshare.com/files/323641732/audioroute_1.2.020.cab

I've used it for both WM to PC and WM to landline, the connection to the PC was much better but I haven't have significant delay when calling landlines. I find that it can vary quite a bit, it probably depends on the amount of people online as well as other factors. On the rare occasion when I had a little delay, I hung up, called again and it was gone. Most of these calls were from UK to HK.
Another thing that annoyed me was the fact that the screen was still active during the call and as I was holding it to my face it pressed random things. If you try to press the END key it will turn off the screen as well as the Wifi connection (which I was using to make the call). So instead, I solved this by pressing the start menu then locking it.

I use Fring for skype calls mainly because I can use the main front speaker (change this in Fring options), works great.
http://www.fring.com/download/default_WAP.asp

Audionotes 1.32, missing sound after long time recording?

hi folks, my virgin post to this almighty forum is as below...
I'm using an ancient recording software audionotes 1.32 with my HD. all good except that, sometimes after recording (not an in-call one) the other party in a phone call cannot hear my voice no matter incoming or outgoing, i have a clear reception though. a soft reset will bring it back.
I couldn't identify exactly how it happend. after a few tries, here's my guess. it appears after a short duration of recording, say minutes, the phone calls are not affected. however, for a long duration as i used to record a conference, it's glitchy.
I'm pretty sure it's not a rom or device dependent thing because it occurs with my HD and P800/P3600i and various ROMs etc. my setting of audionotes is, screen is not turned off/no phone call auto-recording/recording stored in device
please anyone got a clue or someone proves mine is not an isolated case?
cheers!
Ye

Groove IP Issues

Does anyone else use Groove IP with their Nexus S?
I have (since I upgraded to ICS) had issues with Groove IP, and "growing latency". Basically the latency in my call seems to grow as the call goes on, from barely noticeable to >5 seconds. Opening any other app during the call seems to increase this latency, but even if I don't do anything during the call, this occurs. I can hear popping noise when the call is dropping packets (and the latency is increasing) also.
I do have "keep screen on" enabled, and have tried a bunch of different settings to try and fix this, but nothing works.
Does anyone else have this issue? I emailed the developer, and he sent back some settings recommendations, but none of them helped. I have tried a bunch of different settings also. I'm honestly wondering if it is specific to the Nexus S and ICS at this point.
If anyone has time, and can try Groove IP Lite with their Nexus S, I would be very appreciative. Install the app, and try making a call to 1-909-390-0003 (a free echo test number). Try opening a few apps during the call, or sit the phone down for a few minutes, and then see if the latency increases (the time from when you speak to when you hear yourself).
Thanks in advance!!!

Try talkatone
Sent from my Nexus S

sdhanjal15 said:
Try talkatone
Sent from my Nexus S
Click to expand...
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I have - it has a different issue. About half of my outgoing calls don't connect properly, and appear to connect, but all I hear is silence. The person I am calling, or called me just hears silence.
According to the developers, this is a problem that is occurring for a lot of people, and is on Google's end, even though I do not have this problem with Groove IP and sipdroid.

I made a couple of calls. It works
Sent from my Nexus S

i have no problems with takatone too

Are both of you using ICS?

xur17 said:
Are both of you using ICS?
Click to expand...
Click to collapse
Yes I am
Sent from my Nexus S using Tapatalk 2

Yupp
Sent from my Nexus S

do you realize ICS has a built-in SIP?
download and install sipdroid
it will have you set up an account using your google voice login.
after you set it up, you can actually uninstall sipdroid
then, go to pbxes.com
log in and click "extensions" on the left, then click <Sipdroid-200> that shows up under it. create a password and click submit.
then click "personal data" on the left. fill this out and create a password in the given field.
now, go to your dialer
Push menu>settings and scroll all the way down to "Internet Call Settings"
Choose "Ask for each call" under "Use Internet calling"
Tap "accounts"
Check "Receive incoming calls", then tap "Add Account" at the bottom
the info to fill is as follows:
Username: name-200 (it will be the username listed under the extensions page. generally it's formatted like "firstnamelastname", no caps. don't forget the "-200" part)
Password: the password you set up in the extensions section
Server: pbxes.org (yes, .org, not .com like you visited earlier)
Under "Optional settings", Transport type: TCP
now back out. If you did it right, under SIP Accounts, it should show:
[email protected]
Primary account. Receiving calls
now whenever you dial a number on the ICS stock dialer, it will ask you if you want to make the call over the internet or via cellular service. also, if anyone calls your google voice number, your dialer will ring as normal. it will show "Internet Call" under the caller's name on the caller id if it's coming from google voice

I tried using the stock sip client, but the latency is not very good, and there is no way to configure the codecs that it uses. I've honestly been pretty happy with sipdroid, other than the fact that it doesn't look all that great, and is a little buggy.
I have found sipdroid's latency to be better than any other sip client, including csipsimple.

gotta try native
Sent from my DROIDX using Tapatalk 2

Gnex GSM with JB here.
I set up native internet calling with SIP using the guide above (thanks!). However, it only seems to recognize certain numbers. Most of the time it will not ask whether I want to use internet or provider, and just use my provider.
I can add contacts' numbers to a new field called "internet call" which seems to work if I choose it when calling, but this is several extra steps I'd like to avoid. Why won't it use internet calling for all numbers?

I contacted the GrooveIP developers (SNRB Labs). After trying some options, we finally found out that increasing the "Speaker Buffer" in the troubleshooting section from small to large drastically improved the problem of increasing delay during a call.
Also, in the process, I found out that there is a decent way to instigate the delay problem. Start a call to the echo test number 1 909 390 0003, make a short test noise ("test" works), time the delay, open YouTube and play a high quality video for a minute or so (or run a speed test), make the same short test noise and time the delay. With the speaker buffer set to small, for me the delay increased from about 1 second to 2.5 seconds. With the speaker buffer set to large, for the me the delay barely increased at all (maybe 1 to 1.2 seconds, but that's not an exact measurement).

VOIP on the Nexus 5

My fiancée and I got ourselves a pair of Nexus 5's a couple of weeks ago, switched from Verizon to the T-Mo $30 100/Unlimited/5GB(Unlimited) plan and are now happily saving over $100 a month in the process. Yay us!
However, I've been struggling with VOIP with varying degrees of success. I've spent a considerable amount of time researching and configuring and tweaking, and I'd like to share my findings, as well as get some feedback on some things I may have missed.
One of the first things I tried was the Google Voice/PBXes/CSipSimple method, which produced terrible call quality. Everything from echo to background noise. No matter what I did (and believe me, I tried everything I could find) the call quality was just terrible. Changing the mic source, enabling mode audio API, changing the SIP audio mode, changing codecs, nothing really helped. Battery life was great, but the call quality was pretty much unusable. I could hear myself echoing, the other party could hear their own voice echoing, and/or there'd be too much background noise, or I'd be too quiet, etc.
Next, I tried Talkatone (paid for premium). Connection problems galore! I'd have several "lag fests" over WiFi (never tried it on LTE) even when I was sitting right at the router. Everything would cut out for about 30-45 seconds and then resume as if nothing happened, and this occurred 2-3 times over the course of a 10-15 minute call. Yes, I ruled out a connection/router issue. Battery life was "OK" but it wasn't as good as it was with CSipSimple.
I then tried GrooveIP (paid). Lots of echo here. Again, no setting or combination of settings really seemed to get rid of it. Tried as I might, the echo was always there. Battery life was on par with Talkatone.
Next, I decided I'd go a different direction and tried Skype. The voice quality was much improved, with no echo, but complaints of background noise, especially while on speakerphone. This has been passable, though not "ideal" (I know, VOIP isn't perfect). The big issue with Skype has been the absurd battery drain. A 30 minute call drained my battery by almost 20% and Skype was topping the charts by a long shot on the battery usage.
I know there are other options out there such as Viber, but I've not seen a whole lot out of them in terms of reviews, etc. I may just end up trying Viber and seeing how it pans out, but the options are starting to run out.
I know part of the problem is the same one the Nexus 4 had with the microphone(s) but, I'd like to think I just might be overlooking something. If anyone feels they've "solved the problem" please share your settings, as I'm sure I'm not the only one who feels as though they're banging their head against a wall here.

Fenuxx said:
I know there are other options out there such as Viber, but I've not seen a whole lot out of them in terms of reviews, etc. I may just end up trying Viber and seeing how it pans out, but the options are starting to run out.
I know part of the problem is the same one the Nexus 4 had with the microphone(s) but, I'd like to think I just might be overlooking something. If anyone feels they've "solved the problem" please share your settings, as I'm sure I'm not the only one who feels as though they're banging their head against a wall here.
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Viber works well for me and I only hear a slight echo if I'm talking with Nexus 4 users. Give it a shot. Tango might be worth a try, too. Good luck.

Well, I believe Csipsimple is the best voip client available. So, you'll most likely want to go back to your first solution, but replace pbxes with Callcentric, voip.ms or another voip provider. I've tried everything you did as well (plus a few more options) and with the exception of Skype, found the quality to be unacceptable. What I'm suggesting won't be free, but the cost is extremely low. Actually, voip.ms could be a very good solution for you. You would establish and fund one "account", but set up separate "sub-accounts" for yourself and your fiance. If you wanted to use GV exclusively, you could then purchase a couple of DIDs and set up GV to forward to them. I use an app on my phone called Groove Forwarder that changes my GV forwarding settings based on my data connection. If I'm on LTE, etc..., it forwards to my T-Mobile number. When I'm connected to Wi-Fi though, it switches to my Flowroute (another voip provider) number. Also fwiw, you can use voip over LTE if you want. Being in a moving vehicle set up that way will cause issues however.

adrman said:
Well, I believe Csipsimple is the best voip client available. So, you'll most likely want to go back to your first solution, but replace pbxes with Callcentric, voip.ms or another voip provider. I've tried everything you did as well (plus a few more options) and with the exception of Skype, found the quality to be unacceptable. What I'm suggesting won't be free, but the cost is extremely low. Actually, voip.ms could be a very good solution for you. You would establish and fund one "account", but set up separate "sub-accounts" for yourself and your fiance. If you wanted to use GV exclusively, you could then purchase a couple of DIDs and set up GV to forward to them. I use an app on my phone called Groove Forwarder that changes my GV forwarding settings based on my data connection. If I'm on LTE, etc..., it forwards to my T-Mobile number. When I'm connected to Wi-Fi though, it switches to my Flowroute (another voip provider) number. Also fwiw, you can use voip over LTE if you want. Being in a moving vehicle set up that way will cause issues however.
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Yeah, I also tried the Callcentric+PBXes route for the iLBC codec, which didn't seem to help. I'm not entirely convinced it's the PBX provider that's at fault, as I don't have these weird audio issues with CSipSimple+PBXes/Callcentric on my "home phone" (separate Google Voice account) which is an old DROID Incredible 2. Voice quality there is fine.
I did look into voip.ms, but when I signed up (late at night), they forced a "manual authentication" on me (why, I don't know) and I needed to contact support. I tried logging in the following morning, only to be greeted with a message about my IP address not being whiteflagged and not being authorized to access the account. Being that my IP address is dynamic, I don't think I want to constantly fight that battle about "approving" my IP address whenever it changes.

Create a ticket with voip.ms support to inquire. I've only good things to say about their response times and help.

Does anyone have bluetooth headsets working with csipsimple? On my nexus 5 I've yet to find a sip phone that works correctly with a headset.

Fenuxx said:
I then tried GrooveIP (paid). Lots of echo here. Again, no setting or combination of settings really seemed to get rid of it. Tried as I might, the echo was always there. Battery life was on par with Talkatone.
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groove and google voice gave me no echo when calling a landline from my wifi connection. i think this has to be your internet access that would be messing this up. . .or maybe it's just bad for voip to voip calls

I can personally attest to voip.ms + csipsimple + g729 codec ($10 dollars in the playstore) reliability as a voip setup for my Nexus 5. My set up is basically that GV forwards to my voip.ms DID which rings directly to my Nexus 5's csipsimple app. In the event that im not registered in csipsimple (e.g. lose connection, servers go down, etc) I have failover set up w/ voip.ms to ring to my real tmobile phone number. I have zero issues with call quality or echo and I have had full conversations with people on the phone even while driving. I also used this guys tip when first setting up, these may or may not change a thing but Ive had my csipsimple configured with these settings since day 1 also.
1. Go to settings
2. Click the menu button -> Expert Mode
3. Go to “media” -> select echo mode and choose WebRTC (probably already chosen)
4. In “media” go to “Audio troubleshooting” -> “Mic source” -> Voice call
5. in “Audio troubleshooting” -> “Audio implementation” -> Java

I use flowroute + csimpsimple (G729). Call quality is excellent and low latency on WiFi and LTE, and not bad over HSPA/HSPA+.
My main issue at the moment is bluetooth. I cannot get it to work with the bluetooth in my car (only bluetooth I have). I can get incoming audio OK, but it appears to be using the phone microphone for outgoing audio instead of the car microphone and it's very garbled and noisy.

There was a software issue in 4.2 regarding inline mic gain, 4.2.2 fixed it. GroovIP free worked fine for me after the update. There is only a few months left of google voice as they are shutting it down on May 15, 2014.

I have been using Viber for over two years. Works perfectly fine. Try it.
Sent from my Nexus 5 using Tapatalk