sip Questions
5
I get this error when I try to establish a new call from pjsip:
pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=...
3
Solved
Please tell me when a SIP call return 487 Request Terminated?
Is it a termination issue?
5
Solved
2
Solved
I developed a few programs that runs well on Python 3.5.4, but because of some errors about win32 made me go to Python 3.6.4, but when I build my project with pyinstaller, I get:
C:\Users\User\Des...
Wheaton asked 13/7, 2018 at 12:7
4
I am new to MjSip and I use MjUa for creating a client. I want to connect to a asterisk server. it support G.711 but I can not config my app.
I use this config:
media=audio 4000 rtp/avp {audio 0 ...
Toll asked 6/4, 2013 at 14:18
6
I have a trivial doubt with respect to SIP.
I tried googling and referring many books, but still I am not able to find a solid reason for adding from-tag in SIP request.
Example SIP request (Snap...
Ephrem asked 8/10, 2014 at 18:46
2
Solved
I am building an application that gets real-time audio from our organization's VoIP system, records the call and transcribe the real-time voice. The transcription then passed to our analytics engin...
Quincentenary asked 30/1, 2019 at 15:7
5
Solved
I have created a sip trunk from One Asterisk(version 11.2.1) say 'A' server to another Asterisk server(11.7.0) say 'B', and I am getting sip response 200 ok.
But when I start calling on a DID...
2
Solved
Are there reliable open source libraries written in c++ to implement SIP and RTP protocols ? If not , is it easy to implement them using boost.asio ?
Dismissive asked 30/3, 2012 at 1:37
4
Solved
I am very new to RTP , can some one explain about the CSRC and SSRC in general?
From http://www.rfc-editor.org/rfc/rfc3550.txt ,
what it says is : The SSRC field identifies the synchronization sou...
2
Solved
In a SIP video call, the receiver of the video stream respond with the capabilities of its decoder.
The parameter which defines that is the profile-level-id. Here is an example value of the profil...
1
Solved
I'm trying to implement a SIP mid-call mobility using Linphone as UA and Kamailio as SIP proxy/registrar. I start the communication between two UAs in the same network (A) then I move one of the UA...
1
I'm trying to detect the state of an outgoing call when it starts playing ringback tone. I have tried various approaches for detecting this state. Here are some of them:
1. Using PhoneStateListener...
Melba asked 31/7, 2018 at 10:29
3
Solved
This may look like a very simple question, but I haven't found the answer on the Internet.
Anyone can give me clues how to performs a NAPTR query for a SIP domain? (this is mostly for DNS lookup),...
2
I have three doubts that require some clear explanation .
A clear purpose of those two JavaScript SIP library in relation to Webrtc and sip signalling.
Difference between Sip.js and JsSIP Ja...
Maryn asked 4/5, 2018 at 14:22
1
Solved
My goal is to perform a call using VoIP and play an audio file (no matter what format) with Python and record the call. I found some libraries but their documentation is unclear and they don't seem...
1
Solved
Is it possible to open chat window in Microsoft Teams app from a web page?
I am creating one web application that shows list of members in a team. When user double clicks on item in that list, it s...
Frequency asked 20/9, 2019 at 10:5
3
I built a VoIP calling app which maintains a persistent connection with the server to listen to any incoming calls. I implemented a background service to do this.
But since Oreo, this running code...
Hornbeck asked 15/7, 2019 at 8:0
3
Solved
Are there any non-GPL SIP libraries/SDKs that'll let me implement SIP for iOS?
Hermia asked 1/6, 2012 at 16:56
2
I have an interest in APNS and GCM push notifications for SIP VoIP on iOS and Android, respectively.
It would appear that Linphone may support GCM, according to the "Receive data from Internet" pe...
Magnetoelectricity asked 25/1, 2014 at 21:10
1
I am working with android SIP for VoIP. The application is receiving calls successfully. However, initiation of a call is having some bugs.
There is no error in the logs but info says:
" I/art: T...
Defamation asked 9/6, 2016 at 20:32
2
Solved
I am very much new to Kamailio server . I found out that the command to add users is
"./kamctl add ". But how can we find out the number of registered users or how can we know the number of authen...
0
I have web application to make audio calls between clients. As server I use FreeSwitch with SIP (using SIP.js) through secure web socket. When I make call between firefox and firefox everything wor...
Metametabel asked 15/11, 2018 at 18:28
1
Solved
I am trying to add support for bluetooth devices like headsets/headphones/car stereo to my android app which allows user to make SIP/VoIP calls. I am trying to mostly address the requirements of An...
1
Solved
I'm trying to use Twilio to send and receive SMS on my iPhone. I've gotten inbound and outbound calling set up following these instructions. I'd like the equivalent instructions for SMS. I'm presen...
Hurter asked 30/8, 2018 at 15:33
1 Next >
© 2022 - 2024 — McMap. All rights reserved.