sip Questions

2

Solved

Now I test webrtc communicate with SIP Client(sx20) I send invite message with webrtc sdp. but sip client answer has not finger print, and sip client answer is not SRTP just RTP. So I need to t...
Merger asked 13/5, 2014 at 6:43

4

The current API changes for iOS9 state that -setKeepAliveTimeout:handler: is deprecated. Up to now, this was the only way that a VoIP SIP app on iOS could maintain its registration with the SIP-se...
Trachytic asked 10/6, 2015 at 13:53

5

Solved

I know pretty the differences between UDP and TCP in general (eg. http://www.onsip.com/about-voip/sip/udp-versus-tcp-for-voip) Question is, in what circumstances would using TCP as the transport h...
Hutment asked 26/3, 2013 at 18:29

3

I am working on with pjsip for iOS, I have configured the pjsip and able to register and reregister for specific time interval, but there is a scenario where I want to change the pjsip account deta...
Zendah asked 24/4, 2015 at 7:15

7

Solved

I have an android project where I use native code to do stuff with SIP (using libosip2 and libeXosip2). My native code is compiled together with the libraries' sources into one module. The code co...
Neoprene asked 24/8, 2014 at 18:39

1

I am trying to implement a scenario whereby App-User A can make a voice (video not so important now) call to App-User B on Android. It seems difficult to come across a good tutorial that covers the...
Richart asked 19/4, 2016 at 14:21

2

Here is my registration code: protected void initializeManagerOpen(){ consoleWrite("initializeOpen"); if(mSipManager==null) { return; } SipProfile.Builder builder; try { builder = new SipP...
Maracanda asked 15/12, 2013 at 11:43

3

Solved

Greets! I am developing (trying to develop) a VoIP SIP application for Android, and after two weeks of bickering with mjsip, pjsip and the sdk's libraries, I have settled on JAIN-SIP. The librarie...
Bant asked 18/9, 2014 at 9:21

3

I am trying to compile linphone source code. I've downloaded the code from Here. When I started the xcode all the libraries are missing. I have installed all ports specified in README file. I did ...
Reinstate asked 5/4, 2013 at 11:6

4

Solved

I'm looking into implementing a browser-based VOIP solution that uses SIP and WebRTC and that connects to the PTSN. Basically, users give me their SIP credentials and I use WebRTC to acccess ...
Supposition asked 15/12, 2012 at 21:5

1

Solved

what is the difference between ISIM and USIM in VoLTE/LTE/4G scope? What is its use in SIP REGISTRATION process? Why we need two type of it, and does it depend on operator?
Micromho asked 2/2, 2016 at 5:54

1

Solved

What exactly is the difference between a session, a dialog and a transaction? Does all must be present together?
Encircle asked 1/2, 2016 at 14:42

0

I am trying to run the SIPDemo downloaded from Google Andoid sample applications, but not able to get it running. The issue I am facing is that after I invoke SipManager.open passing it a profile ...
Payment asked 3/2, 2016 at 16:14

1

Scenario: I'm using WebRTC (Google's libjingle) on iOS and PeerConnection is setup using a TURN server and I'm waiting for all candidates to gather before I send them to the peer (I'm using SIP). ...
Digestible asked 21/1, 2016 at 15:27

1

I am attempting to setup SIP communication with an internal server (using the PJSIP library), however, this server requires a custom header field with a specified header value for the REGISTRATION ...
Daily asked 11/11, 2011 at 17:46

1

I am trying to make an outbound call from a standalone UCMA application to a SIP provider (Gamma) which authenticates based on connecting IP address. Here is the code with which I am trying to achi...
Haydenhaydn asked 25/6, 2015 at 13:38

2

Solved

I've installed a default out of the box FreeSwitch instance but when I try to make an internal call (extension to extension) it take around 12 seconds before the call is established and I can hear ...
Sato asked 26/9, 2015 at 17:14

3

Solved

I need to capture SIP and RTP traffic to find a problem with something. I can capture SIP messages fine but am having a problem with capturing the RTP traffic. I've tried the following but this i...
Salivate asked 27/7, 2012 at 10:22

5

I need to create use an SIP stack on Android, which will work with asterix and will give users the possibility to change codecs (i need to implement G729 and some other codecs). I'm new in th...
Mammy asked 20/3, 2012 at 13:18

1

Solved

For a server which tries implement video chatting or (multimedia or text chatting for that matter) using RTP which one should be used for control? SIP or RTSP? I went through the abstract of the co...
Capricecapricious asked 13/9, 2015 at 1:46

3

Solved

I am building an RTC iOS app client. I am using the google WebRTC iOS library. However, since WebRTC doesn't implement signalling I am searching for an easy way to implement a SIP stack at the sign...
Hilarius asked 20/8, 2015 at 12:13

2

There are about 20 questions on Linphone ios build in StackOverflow. A couple of them ask about integrating Linphone to an existing xcode project. I have followed them all. I am able to build and r...
Judithjuditha asked 6/8, 2013 at 13:9

2

Solved

Can anyone please clarify - if both, contact and from header, contains the address of originator of SIP INVITE request, then what is the difference between them?
sip
Quadriga asked 24/6, 2015 at 18:39

5

Solved

I need to develop the VOIP application between 2 android devices. As I know there is a SIP protocol used for this purpose but it requires registation to SIP server and access to internet for SIP si...
Address asked 24/6, 2012 at 11:17

1

Solved

Is there any configuration to enable VOIP on device? Or these methods really show that my device does not support VOIP?
Greenwood asked 18/10, 2011 at 13:33

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