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  • SIP UAS asks for OPTIONS

    - by TacB0sS
    Hey, I have UAC that registers to a UAS, after registration the UAS sends me an OPTIONS request, what should I answer it? only the audio media streams? Update I: Allow me to explain myself better... if I want to invite someone to a session I USE the INVITE method and negotiate the media then, for that specific session. But once I register to the server, and it asks me for OPTIONS, then what should I supply, everything my client supports? once I answer it would it deduce that every INVITE I would request from now on would use these medias? or would I need to supply new media with every request? Update II: Hi Wiz, I was in the process of building a negotiation system, so i tried it out and replied the UAS here is the sort dialog we had: OPTIONS sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK45b197cb;rport=5060;received=xx.xx.xx.xx From: "Unknown" <sip:[email protected]>;tag=as66cf26df To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Freeswitch 1.2.3 Max-Forwards: 70 Date: Sat, 05 Jun 2010 12:06:43 GMT Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 OPTIONS In Response To 102: SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bK45b197cb;rport=5060;received=xx.xx.xx.xx From: "Unknown" <sip:[email protected]>;tag=as66cf26df To: <sip:[email protected]> CSeq: 102 OPTIONS Call-ID: [email protected] Allow: INVITE,CANCEL,ACK,BYE,OPTIONS Content-Type: application/sdp Content-Length: 248 v=0 o=310 4515233118481497946 4515233118481497946 IN IP4 10.0.0.1 s=- i=Nu-Art Software - TacB0sS VoIP information c=IN IP4 10.0.0.1 m=audio 40000 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 This response caused the server to stop sending me the options request, does this means I can only use these parameters with the server now? or as you said, it does not matter? Thanks, Adam.

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  • Getting RINGING response on SIP UAC without sending it from the other UAC

    - by TacB0sS
    Hi, I hope this would be my last question about this SIP subject, I have managed to overcome the last issue I had by asking a friend to help me from a remote computer, I'm able to connect between the computers, but here is the thing, according to all the examples I saw, the Callee should invoke the Ringing response, but in my application case I didn't implement it yet, but I still receive on the Caller UAC a Ringing response, this is the SIP messages that are on the caller end: Outgoing Request 5: INVITE sip:[email protected] SIP/2.0 Contact: "Client 310" <sip:[email protected]> From: "Client 310" <sip:[email protected]> Max-Forwards: 32 CSeq: 2 INVITE Call-ID: [email protected] Allow: INVITE,CANCEL,ACK,BYE,OPTIONS Content-Type: application/sdp Proxy-Authorization: Digest username="310",nonce="012afffb",realm="asterisk",uri="sip:[email protected]",algorithm=MD5,response="d19ca5b98450b4be7bd4045edb8a3a2f" Via: SIP/2.0/UDP hostName.hn:5060 To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Content-Length: 257 v=0 o=310 7108915969559970847 7108915969559970847 IN IP4 xxx.xxx.x.xxx s=- i=Nu-Art Software - TacB0sS VoIP information c=IN IP4 xxx.xxx.x.xxx m=audio 3312 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 Incoming Response 6: SIP/2.0 100 Trying Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 Incoming Response 7: SIP/2.0 180 Ringing Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Length: 0 Call to: [email protected] is Ringing Incoming Response 8: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: <sip:[email protected]> Content-Type: application/sdp Content-Length: 264 v=0 o=root 27669 27669 IN IP4 yy.yy.yy.yy s=session c=IN IP4 yy.yy.yy.yy t=0 0 m=audio 10914 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Incoming Response 9: SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP hostName.hn:5060;branch=f8d171d3278788df9e03eb9cf3acba70-xxx.xxx.x.xxx-2-invite-hostName.hn-5060333732;received=79.181.6.233 From: "Client 310" <sip:[email protected]> To: "Client 320" <sip:[email protected]>;tag=as5a8fa200 Call-ID: [email protected] CSeq: 2 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 I do not respond to the invite, that is why all this is happening, but why am I getting a ringing if I'm not the one sending it. Thanks, Adam.

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  • Router gets disconnected once I terminate my SIP application

    - by TacB0sS
    Hey, Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller end I see the Ringing response... now here is interesting part, if I close my application with out any notification to the server my router disconnects and restart, after a short while (30 - 150 sec). I could fix that if I would complete the ACK BYE process, but I'm just wondering why does my router hangs up? any ideas? My Router is TNN-Siemens SL2-141, thought this might matter Update: this is what I found: SIP ALG allows two or more simultaneous VoIP phone calls made by VoIP clients through this router. which means that if I disable it I would not be able to do the testing I'm trying so badly to do, and since I don't have access to another router, I must handle it with the bug then... I can say that this never happened to me with one user connecting, but then again I didn't have anyone to invite then, I received from the SIP UAS 503 when I tried to invite an imaginary user. This bug only occur after I connected the second SIP UAC and invited it and closed the application. Adam.

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  • Unregister SIP UAC message

    - by TacB0sS
    Hi, I've looked so much on the internet, but I could not find a any SIP Unregister example, and when I search RFC 3261,3665 the word does not even appear, perhaps I'm searching for the wrong phrase. I manage to understand the part of setting the expires to zero, but it still does not work and I could not find documentation about how a formal unregister should be. Does anyone knows how to compose an Unregister SIP Request? or what should I search for it? Thanks in advance, Adam Zehavi.

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  • process of connecting RTP with SIP via SDP & land lines

    - by TacB0sS
    Hello to everyone, I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is working fine registers to server invites, and authenticates. I didn't complete any calls to clients yet because of the content header needs to be filled up (which I didn't yet so I get a 503 from the server which is OK I guess). for a long time I didn't know where to start with the media session, and slowly learned how to use the JMF and I've constructed an object that handles RTP transmitting, now I'm standing at the cross road, on the one hand I have my SIP signaling but it needs the SDP content header to complete the invite, and on the other I have the RTP which is knows how to p2p. For me to complete my design I require your help with the following questions: Is there an easy//a simple//an implemented way to convert the Audio/Video format from the JMF into SDP media headers? or even a generator that I would input all the parameters for the content header, and it would generate a content header fast, or do I have to implement this myself? Once I've finished constructing the SDK and the SIP is up and running and I get an OK response from the server (after ringing and all), how do I start the media session? do I connect p2p according to caller details I send in the SIP invite? If 2 is correct, then how does a connection to land lines would be? does land lines knows that once they send an OK back to server they listen/start RTP session on a specific port? Or did I get everything wrong? :-/ I really appreciate any help I could I get, I looked every where for answers but they are not clear, they ignore question 2 as if it was an obvious thing, but for me it just isn't. Thank in advance, Adam Zehavi.

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  • JMF RTPManager transmitting side

    - by TacB0sS
    I was wondering please, the RTP manager in the JMF can perform as a uni-cast,multi-cast, uni-multi-cast, if the session is multi cast the you add the local address to the target list, why is that? what is the logic and effect behind this? thanks for your help, Adam.

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  • SDP media field format

    - by TacB0sS
    Hey, I would like to create a SDP media field with its attributes, and there are a few things I don't understand. I've skimmed and read the relevant RFC and I understand most of what each field means, but what I don't understand is how do I derive from the Audio/Video Format of the JMF, which parameters of the format compose the rtpmap registry entries I need to use. I see many times the fields m=audio 12548 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv these are received from the pbx server I'm connecting to, what do they mean in the terms of the JMF audio format properties. (I do understand these are standard audio format commonly used in telecommunication) UPDATE: I was more wondering about the format parameter '0 8 101' at the end of m=audio 12548 RTP/AVP 0 8 101 Thanks in advance, Adam Zehavi.

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  • JMF microphone volume controller

    - by TacB0sS
    How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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  • Loading Native library to external Package in Eclipse not working. is it a Bug?

    - by TacB0sS
    I was about to report a but to Eclipse, but I was thinking to give this a chance here first: If I add an external package, the application cannot find the referenced native library, except in the case specified at the below: If my workspace consists of a single project, and I import an external package 'EX_package.jar' from a folder outside of the project folder, I can assign a folder to the native library location via: mouse over package - right click - properties - Native Library - Enter your folder. This does not work. In runtime the application does not load the library, System.mapLibraryName(Path) also does not work. Further more, if I create a User Library, and add the package to it and define a folder for the native library it still does not. If it works for you then I have a major bug since it does not work on my computer I test this in any combination I could think of, including adding the path to the windows PATH parameter, and so many other ways I can't even start to remember, nothing worked, I played with this for hours and had a colleague try to assist me, but we both came up empty. Further more, if I have a main project that is dependent on few other projects in my workspace, and they all need to use the same 'EX_package.jar' I MUST supply a HARD COPY INTO EACH OF THEM, it will ONLY (I can't stress the ONLYNESS, I got freaked out by this) work if I have a hard copy of the package in ALL of the project folders that the main project has a dependency on, and ONLY if I configure the Native path in each of them!! This also didn't do the trick. please tell me there is a solution to this, this drives me nuts... Thanks, Adam Zehavi.

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  • Connecting Android device to multiple Bluetooth serial embedded peers

    - by TacB0sS
    I'm trying to find a solution for this setup: I have a single Android device, which I would like to connect to multiple serial embedded devices... And here is the thing, using the "Normal" way to retrieve the Bluetooth socket, doesn't work on all devices, and while it does, I can connect to multiple devices, and send and receive data to and from multiple devices. public final synchronized void connect() throws ConnectionException { if (socket != null) throw new IllegalStateException("Error socket is not null!!"); connecting = true; lastException = null; lastPacket = null; lastHeartBeatReceivedAt = 0; log.setLength(0); try { socket = fetchBT_Socket_Normal(); connectToSocket(socket); listenForIncomingSPP_Packets(); connecting = false; return; } catch (Exception e) { socket = null; logError(e); } try { socket = fetchBT_Socket_Workaround(); connectToSocket(socket); listenForIncomingSPP_Packets(); connecting = false; return; } catch (Exception e) { socket = null; logError(e); } connecting = false; if (socket == null) throw new ConnectionException("Error creating RFcomm socket for" + this); } private BluetoothSocket fetchBT_Socket_Normal() throws Exception { /* The getType() is a hex 0xXXXX value agreed between peers --- this is the key (in my case) to multiple connections in the "Normal" way */ String uuid = getType() + "1101-0000-1000-8000-00805F9B34FB"; try { logDebug("Fetching BT RFcomm Socket standard for UUID: " + uuid + "..."); socket = btDevice.createRfcommSocketToServiceRecord(UUID.fromString(uuid)); return socket; } catch (Exception e) { logError(e); throw e; } } private BluetoothSocket fetchBT_Socket_Workaround() throws Exception { Method m; int connectionIndex = 1; try { logDebug("Fetching BT RFcomm Socket workaround index " + connectionIndex + "..."); m = btDevice.getClass().getMethod("createRfcommSocket", new Class[]{int.class}); socket = (BluetoothSocket) m.invoke(btDevice, connectionIndex); return socket; } catch (Exception e1) { logError(e1); throw e1; } } private void connectToSocket(BluetoothSocket socket) throws ConnectionException { try { socket.connect(); } catch (IOException e) { try { socket.close(); } catch (IOException e1) { logError("Error while closing socket", e1); } finally { socket = null; } throw new ConnectionException("Error connecting to socket with" + this, e); } } And here is the thing, while on phones which the "Normal" way doesn't work, the "Workaround" way provides a solution for a single connection. I've searched far and wide, but came up with zip. The problem with the workaround is mentioned in the last link, both connection uses the same port, which in my case, causes a block, where both of the embedded devices can actually send data, that is not been processed on the Android, while both embedded devices can receive data sent from the Android. Did anyone handle this before? There is a bit more reference here, UPDATE: Following this (that I posted earlier) I wanted to give the mPort a chance, and perhaps to see other port indices, and how other devices manage them, and I found out the the fields in the BluetoothSocket object are different while it is the same class FQN in both cases: Detils from an HTC Vivid 2.3.4, uses the "workaround" Technic: The Socket class type is: [android.bluetooth.BluetoothSocket] mSocket BluetoothSocket (id=830008629928) EADDRINUSE 98 EBADFD 77 MAX_RFCOMM_CHANNEL 30 TAG "BluetoothSocket" (id=830002722432) TYPE_L2CAP 3 TYPE_RFCOMM 1 TYPE_SCO 2 mAddress "64:9C:8E:DC:56:9A" (id=830008516328) mAuth true mClosed false mClosing AtomicBoolean (id=830007851600) mDevice BluetoothDevice (id=830007854256) mEncrypt true mInputStream BluetoothInputStream (id=830008688856) mLock ReentrantReadWriteLock (id=830008629992) mOutputStream BluetoothOutputStream (id=830008430536) **mPort 1** mSdp null mSocketData 3923880 mType 1 Detils from an LG-P925 2.2.2, uses the "normal" Technic: The Socket class type is: [android.bluetooth.BluetoothSocket] mSocket BluetoothSocket (id=830105532880) EADDRINUSE 98 EBADFD 77 MAX_RFCOMM_CHANNEL 30 TAG "BluetoothSocket" (id=830002668088) TYPE_L2CAP 3 TYPE_RFCOMM 1 TYPE_SCO 2 mAccepted false mAddress "64:9C:8E:B9:3F:77" (id=830105544600) mAuth true mClosed false mConnected ConditionVariable (id=830105533144) mDevice BluetoothDevice (id=830105349488) mEncrypt true mInputStream BluetoothInputStream (id=830105532952) mLock ReentrantReadWriteLock (id=830105532984) mOutputStream BluetoothOutputStream (id=830105532968) mPortName "" (id=830002606256) mSocketData 0 mSppPort BluetoothSppPort (id=830105533160) mType 1 mUuid ParcelUuid (id=830105714176) Anyone have some insight...

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  • Exporting Eclipse project with a reference to native library

    - by TacB0sS
    I have an Eclipse project, that uses JMF, I found out I could skip the JMF installation process and still to use the CaptureDeviceManager of the JMF, and to receive the list of devices if I could point my project to the native lib of the JMF. I've managed to add the native lib to the IDE run/debug, but once I export the application to an external runnable Jar, the application cannot find the native lib. the files are located in c:\JMF*.dll I tried to add the folder path to the environment variable in windows - didn't work. I tried to add them into another Jar and add it to the project - didn't work. I tried to add the files into the project - didn't work. I tried to add the path to the class path - didn't work. I tried to add the path to the library path - didn't work. does someone have any sort of a solution? Thanks in advance, Adam Zehavi.

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  • router gets disconnected once I terminate my SIP application

    - by TacB0sS
    Hey, Here is an interesting one, I have a SIP VoIP application which is able to register to the PBX server, and I can invite and see the user call on the callee end receiving an Invite, and on the caller end I see the Ringing response... now here is interesting part, if I close my application with out any notification to the server my router disconnects and restart, after a short while (30 - 150 sec). I could fix that if I would complete the ACK BYE process, but I'm just wondering why does my router hangs up? any ideas?

    Read the article

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