sip

问题 Is it possible to use an Android phone as a simple GSM gateway? The phone would receive a VoiP call using (preferably) Android built-in SIP stack, initiate a GSM call, and bridge audio both ways. After one call is terminated, the other one ends, too. How could I approach the problem? My earlier attempts failed at bridging audio between connections. Is there a SDK supported way of doing this, that I missed? Or do I need to implement some sort of a workaround? 回答1: There are two problems with

问题 Is there any way to make call by just dialing a local IP address? Simply an IP to IP call. How can I do this? What changes should I make in pjsip code? I don't want to register in any server or VOIP provider. The call will happen only in local WiFi, just like SJPhone applications do in Mac (http://www.sjlabs.com/sjp.html). 回答1: You can happily make LAN-only calls with SIP - it is primarily a peer-to-peer protocol, after all. It's a standard part of SIP, in other words. It looks like pjsip

问题 I have created sip application using native sip in android .In it I am getting problem in unregistering account from sip server and every time I am getting DATA_CONNECTION_LOST .I also see in android docs ,but there is no brief explanation of this error.Also it facing various erros while regiterin like in_progress ,transaction_terminated etc which are not having proper explainations in doc . Here is my code : import android.app.Activity; import android.app.Notification; import android.app

问题 I am looking for the SIP stacks for Android and iPhone. I found plenty of similar questions, which are sometimes quite old... I do not care too much if the solution is commercial (but this is preferred) or open source. So far I found Commercial solution from RADVISION for Android Open source SIPHONE for iPhone LINPHONE which covers both platforms. Gingerbreadhas built-in SIP stack - but seems that it has some limitations and might be removed by MNO... My questions are: Does anybody has good

问题 I wanna make SIP peer to peer VoIP call in android. This call should be made in local network without any need to access to internet. In fact I don't want to use any external server to register the profiles. But I don't know what to write in serverDomain in profile builder. This is the constructor public SipProfile.Builder (String username, String serverDomain) Suppose that one ip is 192.168.1.2 and the other is 192.168.1.3 So in initializing function what should I pass as serverDomain to the

问题 I am trying to run test VoIP program as given in http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm I am trying public SIP servers present at http://www1.cs.columbia.edu/sip/servers.html But I always get error message as 14:33:25.515 pjsua_acc.c SIP registration failed, status=408 (Request Timeout) I never used SIP before, I am not able to guess where the problem is. Is there any simple way to test SIP servers? Does anybody know public free SIP server that works?

问题 I'm trying to learn to develop SIP client for Windows Phone 8.1 (following this example). The example mentions to call: Windows.Phone.Networking.Voip.VoipCallCoordinator.RequestNewOutgoingCall() to perform outgoing call. However, from the documentation, RequestNewOutgoingCall is to "Notifies the system of a new outgoing call." Not to connect to the actual SIP server. Anyone knows how to connect to the desired SIP server? Thanks. 回答1: Microsoft does not provide ready solutions for SIP call.

问题 I am getting following error while building pjsip as mentioned in this link. When I run the make command: In file included from /usr/lib/gcc/x86_64-linux-gnu/5/include/errno.h:28:0, from ../include/pj/compat/socket.h:131, from ../src/pj/ioqueue_select.c:38: ../src/pj/ioqueue_common_abs.c: In function ‘pj_ioqueue_recv’: ../include/pj/compat/os_auto.h:149:31: error: ‘EAGAIN’ undeclared (first use in this function) #define PJ_BLOCKING_ERROR_VAL EAGAIN ^ Edit: OS Name: Ubuntu 16.04 LTS && 64 bit

问题 We have a soft phone that's dialing out, on a SIP trunk, through our Asterisk server. The soft phone is sending X Headers that we want to send on to the destination. We see the headers coming into Asterisk, but not going out. Is there something we can do to forward the headers along to the destination? 回答1: Asterisk is no SIP proxy but a B2BUA. This actually means, that it is not forwarding the original request. The call from your softphone gets terminated on Asterisk. Asterisk starts a

问题 I'm developing a Java application based on JAIN SIP with the NIST implementation and would like to enable/view SIP stack debugging. I can't find a working way to achieve that - any help would be appreciated. Thanks! Joe 回答1: Try passing the following properties when you initialize the stack gov.nist.javax.sip.LOG_MESSAGE_CONTENT=true gov.nist.javax.sip.TRACE_LEVEL=32 gov.nist.javax.sip.DEBUG_LOG=logs/mss-jsip-debuglog.txt gov.nist.javax.sip.SERVER_LOG=logs/mss-jsip-messages.xml 回答2: Try this