I want to implement a WebRTC application to be able to make calls over VoIP. My client is running the SIPml5 and in the server side I have installed and confiured the asterisk.
Asterisk was tested with softphone.
Please help??
You will have to configure asterisk with SRTP support.
The below guide will help you
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
Also in the function createSipStack(), place the value of realm, impu and websocket_proxy_url matching with the values.
Let me know if you find any difficulty.
Thanks
Anil
Related
I followed GATT_Server example and advertisment example provided in BlueZ and built a GATT server in Python and it works fine.
I wanted to setup GATT server to use Random address instead of public address. adapter-api and device-api has AddressType field but it is read-only and I can't change it to Random.
Is it possible to setup BlueZ GATT server with Random address?
I think I was having a similar problem. Here's how I solved my problem:
After a lot of searching, I found that PyBluez just uses gattlib for all ble stuff: documentation
I was able to connect to my device with the following: (python 2.7)
import gattlib
requester = gattlib.GATTRequester('EF:76:B5:CC:36:A0', False)
reqer.connect(True, channel_type='random')
I know it's not the same as your problem, but I'm hoping it might help anyways.
I'm new to Mobicents/Restcomm. We need to setup a basic DCCA client application which will receive REST request from clients and send CCR (Event Charging without reservation) to Billing server.
Where can I find some detailed instructions (preferably step by step guide) to start, install, configure, develop/customize and test?
Any help would be really appreciated.
Regards,
Arif
Did you try looking at the documentation at http://documentation.telestax.com/core/diameter/Diameter_User_Guide.html and some of the examples in the project itself https://github.com/RestComm/jdiameter/tree/master/examples ?
You can also look at examples from SIP Servlets using jDiameter https://github.com/RestComm/sip-servlets/tree/master/sip-servlets-examples/diameter-event-charging
As I am trying to install new imc module to my kamailio sip server, I also followed the documentation of it. But unfortunately, I could not run kamailio sip server.
The error which is coming while running it is on this line:
modparam("imc","db_mysql","DBURL")
Where
DBURL = "mysql://kamailio:kamailiorw#localhost/kamailio"
The errors are like syntax error and invalid arguments
What could be the possible chances of this kind of error?
In last what can be the best imc testing software or client?
Please help.
You've set db_mysql instead of db_url while defining a module parameters. please change it to.
modparam("imc","db_url", DBURL)
Also make sure you're loading this module after db_mysql and tm module.
http://www.kamailio.org/docs/modules/3.2.x/modules_k/imc.html#idp1754096
And as it follows the architecture of IRC channels you can use many IRC chat clients to connect such as Pidgin, mIRC and Icechat for Windows, Limechat for Mac OS X, Irssi, WeeChat or XChat for Unix-like systems, ChatZilla for Mozilla browsers or any other standard IRC program.
Registered the users Bob , alice and hunt on local installation of kamailio server on linux redhat . The sipml5 is able to register all the three users and make calls as well as send instant message over sip .
The bad functionality of Call transfer and call hold-resume have worn out my energies .
So far I have tested and tried Sipml5 client with Officesip (without webrtc2sip) , sip2sip.info (without webrtc2sip) , opencloud Rhino (with webrtc2sip) and Kamalio (without webrtc2sip) , but the problem of error generation on sip reinvite persists .
Any help / hint to make Call transfer and call hold-resume work in sipml5 will be great.
ps: I have already posted the questions on webrtc and sipml5 google groups and issue sites however no helpful response as yet .
You could try using the native AMI. I also had the same problems trying to make hold and transfer work but eventually I gave up, implemented them using PAMI ( Php AMI ) + Ajax
Here's the link. I hope this helps
http://marcelog.github.io/PAMI/
I have tried for days now trying to find the right version of red5phone, but to no avail. Also, I need red5 and asterisk to be on different servers. I have followed all instructions described in the various tutorials on web, but nothing helps. I have downloaded 2 different versions of red5phone: sip1, sip_47 from the red5phone google code site, but none of them worked!
When I use:
a) sip1 - shows the correct parameters being passed on the red5 server console, but connection is stuck up with the console displaying the following error:
[NioProcessor-1] ERROR o.r.server.service.ServiceInvoker - Method login with parameters [<sip user>, <sip user>,<sip user>,<sip user pwd>, <asterisk server i/p>, <asterisk server i/p>] not found in org.red5.server.webapp.sip.Application#2d0c94a7
b) sip47 - when I type in the values in the flex interface and check the red5 server console, I see all parameters correct(the ones I passed) except for asterisk server ip that i m trying to connect. Instead, it shows 127.0.0.1 by default and completely ignores the passed ip and regsitration fails.
I am using:
red5 server version: 0.9.1, centos: 4.8(final), red5phone used: sip_47 (tried sip1 as well)
As a desperate measure I tried debugging the source code for red5phone (java and flex files) myself. but when I try to create the environment on my local system, I get several compile errors for missing java packages such as javax.media, org.slf4j, org.red5, etc. really confused and desperate for some guidance. Any tips highly appreciated.
Sunil, I'm also new to this, I would try red5-voicebridge installed in your red5 server.
red5-voicebridge Please let me know if you got it to work.