Has anyone made a small basic application using SIP for android? I have tried using CSipSimple app, with SIP account on antisip.com and sip2sip.com but none of them register properly, and it gives a timeout error.
I also tried those accounts with the SipDemo given with Android, and also in Linphone app. Linphone works fine with a SIP account on Linphone itself, but no other SIP accounts works.
If you want to try it out with CSipSimple, Add Account > OSTN
Account name: OSTN
Username:<yourusername>
Password:<yourpassword>
Server: ostel.co
(Register first #ostel.co)
Or you can download SipWise CE and setup your own local Sip Service Provider - you will not face errors like timeout (as long as you are not far from your wi-fi coverage)
Related
I'm working on a project which is a mobile application developed with Ionic. The project is split into two parts: a mobile front-end and a back-end. The back-end is performing all the calls to external services and applying business rules, while the front-end only calls my back-end.
One of the services I'm using is a IAM service using Oauth2 protocol. I've implemented the authorization code flow to integrate this service. Until now, I was always using the command ionic serve to run my front-end in a web browser, and everything is going well.
But now, for testing purposes, I need to run my front-end in an android emulator. In order for my front-end to contact my back-end, I'm using the IP 10.0.2.2 (which is the alias of the loopback of my machine where my back-end is running).
The problem is with the redirect of the authentication service. It is not something that I have control on, and it can take weeks or months for the service provider to update my client configuration. When I want to authenticate myself through the app running on emulator, it opens a InAppBrowser to perform the authentication. When I successfully authenticated myself against the service, the redirect URI is http://127.0.0.1:8080/xxxx, and this response is sent to that InAppBrowser.
My question is:
Is there a way to replace the base URL of that response (without changing the client configuration), so that I can send the call response to my back-end ?
NOTE: I've been trying to find a solution online, but I can't find anything on that topic. I suspect that in that matter, I probably lack some vocabulary in order to find what I'm looking for.
Is there a way to
I am looking for an open source sip project to add my android application. I found IMSDroid project, downloaded the source code and build on Android ADT.
But when I install the application on my android phone, I cannot register to sip provider(sip2sip.info). I tried on both 3G and WiFi networks but Sign-in always fails. I followed the instructions at this link https://code.google.com/p/imsdroid/wiki/sip2sip_info but no success
By the way, I downloaded CSipSimple application too. I simply filled the registration settings and it worked! I mean CSipSimple application can register to sip2sip while IMSdroid cannot.
I also tried SIP stack of Android. There is a sample application using sip stack of android, called SIPDemo. I also tried this application but its registration also failed.
What is the issue about? I don't want to use CSipSimple because to me it seems more complex than IMSDroid.
my new task is to implement Voice/Video chat native application in Android. I browsed net for the same and it was written that using Android "SIP" API we can do it. I gone through the documents about SIP.In Implementing using SIP we required SIP address for each user which will be provided by different SIP providers. I also checked the SipDemo sample app provided in samples.
Problem is my app will be used by many Users and they can chat with any others.
So, how can we generate SIP address for each user dynamically and establish communication session ?
or each there any other way of doing this ?
Please Help
If you want to test your android app you can generate a free sip account from AntiSip.
and for generating sip account dynamically you have to configure sip on your server with the help of Asterisk:
Two VoIP software can contact each other directly by using direct SIP URI like username#IP:port.
This can work fine on local networks where the UA (sip user agent) has fix address. Otherwise you will need to use a SIP server (This is the main role of a SIP registrar server: to keep a track of the user locations). There are plenty open source software for this like the above mentioned Asterisk but you can also find a lot of services offering free calls (free from softphone to softphone; they charge only if you wish to make outbound calls to landline or mobile numbers).
There is a Google+ mobile application for Android. To receive all the information it must be using some API.
How I can retrieve this information from my phone and see from where this app is getting data? I'd love to access it myself and see how it works.
You are talking about "reverse engineering" a network protocol.
Full sniffing
Get a market enabled emulator: How to install Android Market App on the emulator?
Get a network sniffer: tcpdump, wireshark, ...
Start to sniff what's going over the wire
http://www.thoughtcrime.org/software/sslstrip/ - an ssl stripping proxy
Inject your ca: http://www.mcbsys.com/techblog/2010/12/android-certificates/
This should, in theory, enable you to sniff any https or plain text connection.
Already available APIs
I would not try to hijack the connection. There is already a contacts API, it's called xmpp. You should be able to use the talk integration to pull your friend list.
Future APIs
You may also want to sign up for the upcoming API.
UPDATE
It appears that the Android client uses (at least partially) XMPP. The regular client requires an open XMPP connection and you can see a "RealTimeChat" in your logcat. It looks like the connection is encrypted because you get a "TLS required" message (the client seems to go through the XMPP connection states). I'm not sure if that's used for client based posts or just for server push. You can see the open connections with the help of "netstat". The connection goes away when google plus terminates.
I'd thus expect the API to be really open once released (or rev.eng).
UPDATE 2 (06. Jul. 2011)
Hangout is build on XMPP/MUC+JINGLE (muc == multiuser chat, jingle is roughly a SIP alternative on top of XMPP). And yes, they'll release the the details of it :-)
UPDATE 3 (06. Jul. 2011)
Multiple XMPP components for gtalk/gplus have been revealed by reading the JS code. It also emphasizes that they have build a great deal of features on XMPP.
They haven't launched the API yet. If there is something you want to build on Google+, they encourage you to signup here: https://services.google.com/fb/forms/plusdevelopers/
The API hasn't been released yet. Heck, the product hasn't even been released yet. Thats like developing a video game for playstation 5.
Yes it is. Its coming "soon" read :: http://www.webpronews.com/api-coming-soon-for-google-plus-2011-07
I have to build a SIP application for my studies, i maked an Asterisk server, testing it with xlite and all is working. However, i wanted to test it in my Android phone(has 2.3 Gingerbread), i installed the SIPDemo app on it, adding the useful informations and nothing happens, it doesn't indicate if you are connected on the server or not. I run the application in the Android emulator and it saids "Registration failed".
Can you please help me? I really have to make this project to have my diploma.
Thank you very much.
from asterisk cli (asterisk -rvvvvvvvvvvv) you should type:
sip set debug on
and then try to register, this will let you debug the situation.
This could be an asterisk configuration problem or wrong SIP registration management on client's side.
If your registration is successful after removing password from asterisk's user then you need to check how SIP handles registration process https://www.rfc-editor.org/rfc/rfc3261#page-56 and stick to those rules with your demo.