Hi dear friend,Hello,
Not yet, it's still under development.
I have always respected your contribution to the B4X community and liked almost 100% of your posts on B4X but i never though you would make a disrespectful comment on our project.
Is this unnecessary and disrespectful remark really needed to be wrote here ?
Thank you,
Saif
Hi Saif,Hello,
Not yet, it's still under development.
the new library also manages the password, you should use the new oneHi, I bought your distribution. I have no experience with Java, but have already programmed in B4A. It is not clear to me which folders I need to use to create an application using the SIP. The distribution has two variants of similar folders:
which one should I choose from me 11V B4A do I need?
- VoIPSIPB4A-NewLibrary
- VoIPSIPB4A
I run VoIPSIPB4A the message appears
View attachment 119072
What does the maven artifact message mean? How to solve this problem?the new library also manages the password, you should use the new one
Hello,What does the maven artifact message mean? How to solve this problem?
If you have about $100,000 USD to invest on the servers and APIs to route DID numbers to your SIP server, I'd be glad to tell you how it can be done absolutely free... We've worked on a number of telecom projects using SIP over the years - other programming languages though. Like IPs, you would need to purchase an entire DID phone number subset... And it comes with a yearly fee. You must be licensed to operate a telecom though, which will incur additional fees.This is completely false
My request, since last April, is unique
Allow the communication of the Voip Sip System project with the public telephone network
On the great usefulness of this module you will surely have seen the approval of other users as well
If it is too difficult for you too, patience, we will wait for whoever knows how to do it
Feel free to work on beautifying the code
If you have about $100,000 USD to invest on the servers and APIs to route DID numbers to your SIP server, I'd be glad to tell you how it can be done absolutely free... We've worked on a number of telecom projects using SIP over the years - other programming languages though. Like IPs, you would need to purchase an entire DID phone number subset... And it comes with a yearly fee. You must be licensed to operate a telecom though, which will incur additional fees.In the US, you would need to get approval from the FCC first also. If you have the money, I'll point you in the right direction... $100,000 will get you setup and access about 25,000 phone numbers.
You can purchase a cheap VPS system for development testing over at ionos.com. They're only about $2 / month and no long term contacts. That will provide you with a static IP as well as a remote server to test calls between any two points on Earth. Then you can test locally, and even ask a friend on the opposite side of the world to join in on the test by giving them a call. Performance should really be based on latency and systems performances thoughI reinstalled SDK and Java. After that, the client VoIP started on the AVD, the Java server started on this PC. How can I check the performance of the entire project? I want to test a voice call from one smartphone to another via the Internet.
1. option - Install the application on two smartphones, and run the server on the PC, but I do not have a static IP address for the server.
2. option - run client application on the AVD in the local network on 2 different PCs. Next step, on the third PC, run the Java server. Try to talk through the Microphone.
If you have about $100,000 USD to invest on the servers and APIs to route DID numbers to your SIP server, I'd be glad to tell you how it can be done absolutely free... We've worked on a number of telecom projects using SIP over the years - other programming languages though. Like IPs, you would need to purchase an entire DID phone number subset... And it comes with a yearly fee. You must be licensed to operate a telecom though, which will incur additional fees.In the US, you would need to get approval from the FCC first also. If you have the money, I'll point you in the right direction... $100,000 will get you setup and access about 25,000 phone numbers.
I don' think so...
amorosik not talking for numbers (DID) but only for APIs -talking about the connection at SIP Servers // Like a VOIP/SIP Router with FXs-phone-input/output, like PAP (linksys-cisco), like the simple one included library at b4a (but ofcourse with more options and for all b4x)...
is just code but ...sure is difficult...some of APIs need money - some of them are free but need to learn from "0" what they need.. need to use many codecs too.. some need to use and JTAPI too (for old pbx devices) and many many other...
ofcourse the cost having for this library - the developer sfsameer is very low-cheap and that he did already are too much/many... but not what amorosik need...
Hope all find the way and create a beautiful library... is not so simple...
I bought VoIP SIP sources to use in my application. I am planning to create an application for consulting my clients. Communication with a consultant, I wanted to do with the help of VoIP technology. Therefore, I hoped that the VoIP SIP distribution kit, from Saif Sameer, would solve all my problems with voice communication, without using additional servers. Now I need to make sure that the code is functional, how can I check the capabilities of the VoIP distribution kit. I think to use clients on smartphones, and run the server on a PC using a static IP address. I was sure that in this version I did not need to connect to other SIP servers.You can purchase a cheap VPS system for development testing over at ionos.com. They're only about $2 / month and no long term contacts. That will provide you with a static IP as well as a remote server to test calls between any two points on Earth. Then you can test locally, and even ask a friend on the opposite side of the world to join in on the test by giving them a call. Performance should really be based on latency and systems performances though
I bought VoIP SIP sources to use in my application. I am planning to create an application for consulting my clients. Communication with a consultant, I wanted to do with the help of VoIP technology. Therefore, I hoped that the VoIP SIP distribution kit, from Saif Sameer, would solve all my problems with voice communication, without using additional servers. Now I need to make sure that the code is functional, how can I check the capabilities of the VoIP distribution kit. I think to use clients on smartphones, and run the server on a PC using a static IP address. I was sure that in this version I did not need to connect to other SIP servers.
Registration procedure, the main problem is why I do not want to use public SIP servers. I was hoping to create an application in which my user makes a registration once, inside my application. After that, he will have the opportunity to talk with the consultant via the Internet. I do not want to force the user to pre-register on a public SIP server through a browser. I was hoping that the VoIP SIP Saif Sameer project would solve this problem.If you try to use the B4A client and the latest B4J client, those with user and password, you will see that they can connect to classic sip pbx like Asterisk or 3Cx
Personally I have tried with FreePbx and they seem to work correctly, at least in the features currently present
Then, try to take the next step, try to register the B4a or B4J client directly on the sip server of the voip phone provider
And you will see that this works too
Well, if this second test works, then it means that the basic tools for communicating with a sip server, in the Voip Sip source code project, are present
So, could you tell me what are the technical difficulties in allowing the B4J server to communicate with a sip server, as well as the B4A and B4J client can do NOW?
Where is the B4J SIP server running (IP)? That is where you will need to point your management console. The bind error is indicating that Java was unable to 'bind' to 192.168.2.17 (is this really where the SIP server is installed?!?) Hmmm... That's an odd address (for general user setup)... Read on...I am successfully starting the server from the MAIN procedure.
View attachment 119243
When I click on the "Start server" button, the server crashes.
View attachment 119245
I have tested on different computers when using the "Start" button is always a problem. What is the problem?
Hello,I am successfully starting the server from the MAIN procedure.
View attachment 119243
When I click on the "Start server" button, the server crashes.
View attachment 119245
I have tested on different computers when using the "Start" button is always a problem. What is the problem?
Hello,Registration procedure, the main problem is why I do not want to use public SIP servers. I was hoping to create an application in which my user makes a registration once, inside my application. After that, he will have the opportunity to talk with the consultant via the Internet. I do not want to force the user to pre-register on a public SIP server through a browser. I was hoping that the VoIP SIP Saif Sameer project would solve this problem.
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