https://docs.min.io/minio/baremetal/replication/enable-server-side-two-way-bucket-replication.html#required-permissions
I follow this guide, when I do "mc admin policy add". I got an err:
mc: <ERROR> Unable to add new policy: conditions are not supported for action s3:GetBucketVersioning.
Then I add '--debug':
mc: <DEBUG> PUT /minio/admin/v3/add-canned-policy?name=ReplicationRemoteUserPolicy.json HTTP/1.1
Host: xxxxx:xxx
User-Agent: MinIO (linux; amd64) madmin-go/0.0.1 mc/RELEASE.2022-07-06T14-54-36Z
Content-Length: 1328
Accept-Encoding: gzip
Authorization: AWS4-HMAC-SHA256 Credential=admin/20220707//s3/aws4_request, SignedHeaders=host;x-amz-content-sha256;x-amz-date, Signature=**REDACTED**
X-Amz-Content-Sha256: 26a5f72146edcd356b967fb84a6b1407418205af1904f9408fd2e85b196c98d1
X-Amz-Date: 20220707T094030Z
mc: <DEBUG> HTTP/1.1 400 Bad Request
Content-Length: 237
Accept-Ranges: bytes
Content-Security-Policy: block-all-mixed-content
Content-Type: application/json
Date: Thu, 07 Jul 2022 09:40:19 GMT
Server: MinIO
Vary: Origin
X-Amz-Request-Id: 16FF82A1BB2425B5
X-Xss-Protection: 1; mode=block
{"Code":"XMinioMalformedIAMPolicy","Message":"conditions are not supported for action s3:GetBucketVersioning","Resource":"/minio/admin/v3/add-canned-policy","RequestId":"16FF82A1BB2425B5","HostId":"3473e3d7-6fef-4358-83e7-f7e333eb8675"}
mc: <DEBUG> Response Time: 3.844982ms
---------START-HTTP---------
PUT /minio/admin/v3/add-canned-policy?name=ReplicationRemoteUserPolicy.json HTTP/1.1
Host: xxxxx:xxxx
User-Agent: MinIO (linux; amd64) madmin-go/0.0.1 mc/RELEASE.2022-07-06T14-54-36Z
Content-Length: 1328
Accept-Encoding: gzip
Authorization: AWS4-HMAC-SHA256 Credential=admin/20220707//s3/aws4_request, SignedHeaders=host;x-amz-content-sha256;x-amz-date, Signature=**REDACTED**
X-Amz-Content-Sha256: 26a5f72146edcd356b967fb84a6b1407418205af1904f9408fd2e85b196c98d1
X-Amz-Date: 20220707T094030Z
HTTP/1.1 400 Bad Request
Content-Length: 237
Accept-Ranges: bytes
Content-Security-Policy: block-all-mixed-content
Content-Type: application/json
Date: Thu, 07 Jul 2022 09:40:19 GMT
Server: MinIO
Vary: Origin
X-Amz-Request-Id: 16FF82A1BB2425B5
X-Xss-Protection: 1; mode=block
{"Code":"XMinioMalformedIAMPolicy","Message":"conditions are not supported for action s3:GetBucketVersioning","Resource":"/minio/admin/v3/add-canned-policy","RequestId":"16FF82A1BB2425B5","HostId":"3473e3d7-6fef-4358-83e7-f7e333eb8675"}
---------END-HTTP---------
mc: <ERROR> Unable to add new policy: conditions are not supported for action s3:GetBucketVersioning
(1) admin-policy-add.go:140 cmd.mainAdminPolicyAdd(..) Tags: [cluster202, ReplicationRemoteUserPolicy.json, /dev/stdin]
(0) admin-policy-add.go:140 cmd.mainAdminPolicyAdd(..)
Commit:81c4a5ad6ee4 | Release-Tag:RELEASE.2022-07-06T14-54-36Z | Host:clone-instance-testv3 | OS:linux | Arch:amd64 | Lang:go1.18.3 | Mem:3.3 MB/17 MB | Heap:3.3 MB/7.7 MB.
How can I fix this?
I used helm chart install minio cluster, the tag of docker images is :RELEASE.2021-02-14T04-01-33Z
How to get Status code like 100 and 180 from response when i call with any number from below response in my console.console output for my call response as below so how to get status code for trying and ringing when i call with nay number.
[1]: https://i.stack.imgur.com/YOTL8.png
also display resonse as below :
browser.js:183 JsSIP:WebSocketInterface received WebSocket message +13s
browser.js:183 JsSIP:Transport received text message:INVITE sip:rpj7...#127.0.0.1:55702;transport=WS SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8088;rport;branch=z9hG4bKPjacdc2b28-994d-49de-b3af-f3f7740afb93;alias
From: "F200" <sip:2...#XXXXX.com>;tag=95bffc6b-992b-48dd-83e1-23a2473e271e
To: <sip:rpj7...#127.0.0.1>
Contact: <sip:aste...#XXXXX.com:5060;transport=ws>
Call-ID: 3290aa94-d410-4bb5-ad10-b72e9a05ce04
CSeq: 3467 INVITE
Allow: OPTIONS, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, MESSAGE, REFER
Supported: 100rel, replaces, norefersub
P-Asserted-Identity: "F200" <sip:2...#XXXXX.com>
Max-Forwards: 70
User-Agent: Asterisk PBX 16.8.0-7.5.1.rl.101706.4465cf5
Content-Type: application/sdp
Content-Length: 935
v=0
...trimmed ordinary-looking incoming SDP
a=mid:audio-0
+13s
browser.js:183 JsSIP:Transport send() +1ms
browser.js:183 JsSIP:Transport sending message:SIP/2.0 100 Trying
... trimmed 100 trying body....
+0ms
browser.js:183 JsSIP:WebSocketInterface send() +6ms
browser.js:183 JsSIP:RTCSession new +3m
browser.js:183 JsSIP:RTCSession init_incoming() +1ms
browser.js:183 JsSIP:Dialog new UAS dialog created with status EARLY +3m
browser.js:183 JsSIP:RTCSession newRTCSession() +0ms
App.js:348 {id: 0.5232762505149955, remoteExt: "800", time: 1591730563324, state: "incoming", session: RTCSession}
browser.js:183 JsSIP:Transport send() +3ms
browser.js:183 JsSIP:Transport sending message:SIP/2.0 180 Ringing
... TRIMMED 180 Ringing body ....
Content-Length: 0
+0ms
browser.js:183 JsSIP:WebSocketInterface send() +3ms
browser.js:183 JsSIP:RTCSession session progress +2ms
browser.js:183 JsSIP:RTCSession emit "progress" +0ms
.... session.answer called here ...
browser.js:183 JsSIP:RTCSession answer() +501ms
browser.js:183 JsSIP:Dialog dialog 3290aa94-d410-4bb5-ad10-b72e9a05ce04sgb1kgc5mc95bffc6b-992b-48dd-83e1-23a2473e271e changed to CONFIRMED state +503ms
browser.js:183 JsSIP:RTCSession emit "peerconnection" +1ms
browser.js:183 JsSIP:WebSocketInterface received WebSocket message +2s
browser.js:183 JsSIP:Transport received text message:
+2s
common.js:114 JsSIP:ERROR:Parser parseMessage() | error parsing first line of SIP message: "" +15s
debug # common.js:114
./node_modules/jssip/lib-es5/Parser.js.exports.parseMessage # Parser.js:29
onTransportData # UA.js:950
_onData # Transport.js:303
_onMessage # WebSocketInterface.js:151
Try this:
this.coolPhone = new JsSIP.UA(configuration);
this.setUAEvent();
this.coolPhone.start();
setUAEvent(ua) {
this.coolPhone.on("newRTCSession", (e) => {
console.log("new rtc session", e);
const session = e.session;
session.on("progress", (message) => {
console.log("progress session ", message.response.status_code);
if (message.response.status_code === 180) //RINGING EXAMPLE
{
//put your code here
}
});
}
I use sipml5 with freeswitch and I need to detect when call should be answered automatically. The only part where I can get it from is SIP Invite message:
recv=INVITE sip:username#IP:50598;transport=ws;intercom=true SIP/2.0
Via: SIP/2.0/WSS IP;branch=z9hG4bKd451.8dc49598935d4ebdf937de014cf1d922.0
From: "Device QuickCall"<sip:NUMBER#DOMAIN>;tag=68rtr6c12v9em
To: <sip:michaltesar2#IP:50598;transport=ws>
Contact: <sip:mod_sofia#IP:11000>
Call-ID: dcd8fb4d69f0850840a743c152f4f7358a21-quickcall
CSeq: 89383073 INVITE
Content-Type: application/sdp
Content-Length: 882
Record-Route: <sip:IP;transport=ws;r2=on;lr=on;ftag=68rtr6c12v9em>
Record-Route: <sip:IP;r2=on;lr=on;ftag=68rtr6c12v9em>
Via: SIP/2.0/UDP 37.157.194.240:11000;rport=11000;received=IP;branch=z9hG4bKSNmDFvya0ceaQ
Max-Forwards: 50
Call-Info: answer-after=0;answer-after=0
User-Agent: 2600hz
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY,PUBLISH,SUBSCRIBE
Supported: path,replaces
Allow-Events: talk,hold,conference,presence,as-feature-event,dialog,line-seize,call-info,sla,include-session-description,presence.winfo,message-summary,refer
Content-Disposition: session
Remote-Party-ID: privacy=off;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1459415113 1459415114 IN IP4 37.157.194.240
s=FreeSWITCH
c=IN IP4 37.157.194.240
t=0 0
a=msid-semantic: WMS W2YlkINCSBwtCldHnD3FYpIuFQW9iaH5
m=audio 23162 RTP/SAVPF 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fingerprint:sha-256 03:8E:7D:14:E6:88:F1:75:55:70:40:E5:7F:07:9F:9F:C5:38:43:59:FB:EF:4D:70:0C:C7:F7:24:FC:7B:54:AB
a=rtcp-mux
a=rtcp:23162 IN IP4 37.157.194.240
a=ssrc:1258116307 cname:2vgd3UFMl25Od8lq
a=ssrc:1258116307 msid:W2YlkINCSBwtCldHnD3FYpIuFQW9iaH5 a0
a=ssrc:1258116307 mslabel:W2YlkINCSBwtCldHnD3FYpIuFQW9iaH5
a=ssrc:1258116307 label:W2YlkINCSBwtCldHnD3FYpIuFQW9iaH5a0
a=ice-ufrag:CfWquvL0by0kyxfq
a=ice-pwd:SmtM6ZoiRjWVi8cKdZ1ykDom
a=candidate:8660741513 1 udp 659136 IP 23162 typ host generation 0
a=candidate:8660741513 2 udp 659136 IP 23162 typ host generation 0
a=ptime:20
My VOIP phone detects it from Call-Info header :
Call-Info: answer-after=0;answer-after=0
Is there any way how to access Call-Info header using sipml5?
I also needed to get a SIP header's value for something similar in a project using SIPml5.
What I did, is a bit of a hack, but it works: all SIP signaling messages are logged to browser console (if the debug level is set to "info").
So I found and changed that debug function in SIPml5 library to receive all the incoming SIP messages (regardless of the debug level). You can find the function, by searching for this: function tsk_utils_log_info.
The new function looks like:
function tsk_utils_log_info(s_msg){
if (s_msg.indexOf('recv=') === 0)
{
CatchWebrtcSignaling(s_msg);
}
common_public.PutToDebugLog(3, 'WRTC, EVENT, ' + s_msg);
if (window.console && (__i_debug_level >= 4)) {
window.console.info(s_msg);
}
}
Now I receive all incoming SIP messages in function CatchWebrtcSignaling(msg) where I can parse the message and get any SIP headers value.
You can either make this change in SIPml5-api.js file or you can download the source code from github, make the change and minify/build SIPml5-api.js, by executing "release.sh" from main directory.
in SIPml.js you have event with 'i_new_call' type
, in this event you have o_message variable, you can iterate over this o_message.ao_headers
var dispatchEvent = function (s_event_type) {
if (s_event_type) {
switch (s_event_type) {
case 'i_new_call':
case 'm_permission_requested':
case 'm_permission_accepted':
case 'm_permission_refused':
{
var oNewEvent = new SIPml.Stack.Event(s_event_type, e);
if (s_event_type == 'i_new_call') {
oNewEvent.o_event.o_message.ao_headers.forEach(function (o_header) {
window.console.error('header name '+ o_header.s_name+ 'value ' + o_header.s_value);
});
}
I'm trying to send a 200 OK response using a Sip Servlet after an Invite request by using request.createResponse(200).
Problem is that the Contact header comes out without the "username" part.
Why is this happening?
Relevant Code:
protected void doInvite(SipServletRequest arg0) throws ServletException, IOException {
SipServletResponse respRinging = arg0.createResponse(180);
respRinging.send();
SipServletResponse respOk = arg0.createResponse(200);
String sdpData = "v=0\r\n" +
"o=- 2 2 IN IP4 192.168.50.18\r\n" +
"s=Testing V1.0\r\n" +
"c=IN IP4 192.168.50.18\r\n" +
"t=0 0\r\n" +
"m=audio 63364 RTP/AVP 18 101\r\n" +
"a=fmtp:18 annexb=yes\r\n" +
"a=fmtp:101 0-15\r\n" +
"a=rtpmap:18 G729/8000\r\n" +
"a=rtpmap:101 telephone-event/8000\r\n" +
"a=sendrecv\r\n";
byte[] contents = sdpData.getBytes();
respOk.setContent(contents, arg0.getContentType());
respOk.send();
}
Created 200 Message:
%dMES%22=SIP/2.0 200 OK
Call-ID: ZDE4MzY3NzAzNjRlN2UwZDBmMGZlNDA1MTNlZWJiOWE.
CSeq: 1 INVITE
From: 6000 <sip:6000#192.168.50.18>;tag=d765eb39
To: 2451 <sip:2451#192.168.50.18>;tag=9020141091761795_local.1330939329531_17_16
Via: SIP/2.0/UDP 192.168.50.201:57332;branch=z9hG4bK-d8754z-cc352300a144627e-1---d8754z-;rport=57332;received=192.168.50.201
Content-Type: application/sdp
Content-Length: 221
Contact: sip:IP-HOST:5060;transport=udp
I expected the Contact URI to be sip:2451#IP-HOST:5060;transport=udp
I imagine that jain-sip is automatically populating the Contact URI. It's a perfectly valid Contact URI - the username is not required.
If you need to specify a particular URI, have you tried respOk.addHeader("Contact", "My sip:special#uri:5060")?
I am developing a SIP controller in Java using the NIST implementation of the JAIN SIP API.
I am having trouble making a call from my SIP controller to a softphone via Asterisk.
If I call the softphone directly (not via Asterisk) using its IP address and port number, everything works fine. The call gets established, the softphone hears the audio (RTP data) I send it, and I can receive the audio that it sends me.
However, when I call the same softphone via Asterisk, the call gets established, and I start to receive RTP data from the softphone (via Asterisk). Now, my send stream takes a little while to set up, but while it is being configured I receive the RTP data from the softphone. The problem is that as soon as my send stream is initialized and starts to transmit RTP data, I stop receiving RTP data from the softphone! The result is that after the call is established, I hear the softphone for half a second or a second at most, and then nothing. At this stage the softphone can hear my outgoing RTP-data, but I cannot hear it.
If I don't start transmitting any RTP data, I keep on receiving RTP data from the softphone. But as soon as I start transmitting, it stops coming!
In case it helps, here is the type of SIP-conversation that establishes the call (>> indicates an outgoing message and << indicates indicates an incoming message):
>> INVITE sip:301#asterisk SIP/2.0
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
CSeq: 1 INVITE
From: <sip:null>;tag=JqbJKA
To: <sip:301#asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK34d24b3f748ac08a5ca46f500f110d38353436
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Route: <sip:10.0.84.30;lr>
Content-Type: application/sdp
Content-Length: 106
v=0
o=- 3515232260 3515232260 IN IP4 10.0.85.3
s=-
c=IN IP4 10.0.85.3
t=0 0
m=audio 42138 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<< SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK34d24b3f748ac08a5ca46f500f110d38353436;received=10.0.85.3
From: <sip:null>;tag=JqbJKA
To: <sip:301#asterisk>;tag=as7077f414
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
CSeq: 1 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:301#10.0.84.30>
Proxy-Authenticate: Digest realm="asterisk",nonce="4a1cbda4"
Content-Length: 0
>> INVITE sip:301#asterisk SIP/2.0
CSeq: 2 INVITE
From: <sip:303#asterisk>;tag=JqbJKA
To: <sip:301#asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Route: <sip:10.0.84.30;lr>
Proxy-Authorization: Digest username="303",realm="asterisk",nonce="4a1cbda4",response="249b2b7d7c0e7b54499c632ba410365c",algorithm=MD5,uri="sip:301#asterisk",nc=00000001
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
Content-Type: application/sdp
Content-Length: 106
v=0
o=- 3515232260 3515232260 IN IP4 10.0.85.3
s=-
c=IN IP4 10.0.85.3
t=0 0
m=audio 42138 RTP/AVP 0
a=rtpmap:0 PCMU/8000`
`<< SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436;received=10.0.85.3
From: <sip:303#asterisk>;tag=JqbJKA
To: <sip:301#asterisk>
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,R EFER,SUBSCRIBE,NOTIFY
Contact: <sip:301#10.0.84.30>
Content-Length: 0
`<< SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436;received=10.0.85.3
From: <sip:303#asterisk>;tag=JqbJKA
To: <sip:301#asterisk>;tag=as00faa25e
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:301#10.0.84.30>
Content-Length: 0`
<< SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKd1870f50e9fbf883b3e64fa3ef75dda9353436;received=10.0.85.3
From: <sip:303#asterisk>;tag=JqbJKA
To: <sip:301#asterisk>;tag=as00faa25e
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
CSeq: 2 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:301#10.0.84.30>
Content-Type: application/sdp
Content-Length: 154
v=0
o=root 2593 2593 IN IP4 10.0.84.30
s=session
c=IN IP4 10.0.84.30
t=0 0
m=audio 10294 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
>> ACK sip:301#10.0.84.30 SIP/2.0
Call-ID: 8b92ba1ca9c922bcd266dce086596ce4#10.0.85.3
CSeq: 2 ACK
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK7e16ebc0de9c6eaf901db0e2e58f495f353436
From: <sip:303#asterisk>;tag=JqbJKA
To: <sip:301#asterisk>;tag=as00faa25e
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Content-Length: 0
Here is the code that sets up the RTP-session. First some declarations:
private RTPManager sessionManager = null;
private Processor processor = null;
private SendStream sendStream;`
The following method is called first:
public void startMedia(String peerIp,int peerPort,int receivePort,String format) throws IOException,MediaException,InvalidSessionAddressException
{
stopMedia();
this.format = format;
RTPSessionMgr rtpSessionMgr = new RTPSessionMgr();
rtpSessionMgr.initSession(new SessionAddress(),null,0.05,0.25);
InetAddress localhost = InetAddress.getLocalHost();
SessionAddress localAddr = new SessionAddress(localhost,receivePort,localhost,receivePort + 1);
InetAddress destAddr = InetAddress.getByName(peerIp);
rtpSessionMgr.startSession(localAddr,localAddr,new SessionAddress(destAddr,peerPort,destAddr,peerPort + 1),null);
sessionManager = rtpSessionMgr;
for (ReceiveStreamListener nextListener : receiveStreamListeners)
sessionManager.addReceiveStreamListener(nextListener);
}
Then, to start playing the sound over RTP, this method is called:
public void transmitSound(DataSource ds) throws NoProcessorException,IOException,UnsupportedFormatException,NotRealizedError
{
stopTransmittingSound();
processor = Manager.createProcessor(ds);
for (ControllerListener nextListener : controllerListeners)
processor.addControllerListener(nextListener);
processor.addControllerListener(myControllerListener);
processor.configure();
}
Here is the controllerUpdate() method of the controller listener:
public void controllerUpdate(ControllerEvent event)
{
if (processor.getState()==Processor.Configured)
{
processor.setContentDescriptor(new ContentDescriptor(ContentDescriptor.RAW_RTP));
processor.getTrackControls()[0].setFormat(new AudioFormat(format,8000,8,1));
processor.realize();
}
else if (processor.getState()==Processor.Realized)
{
try
{
sendStream = sessionManager.createSendStream(processor.getDataOutput(),0);
sendStream.start();
processor.start();
}
catch (IOException e)
{
e.printStackTrace();
}
catch (UnsupportedFormatException e)
{
e.printStackTrace();
}
catch (NotRealizedError e)
{
e.printStackTrace();
}
}
}
This is what basically happens after the ACK is sent:
I create an RTP-session for transmitting and listening.
I start initializing a processor for transmitting RTP.
In the meanwhile I receive lots of RTP-data.
The processor finishes initialization and I start sending RTP-data.
At this stage I stop receiving RTP-data if going through Asterisk. If calling a softphone directly, everything works fine.
Any ideas?
Are you sure your handling for send part of RTP is correct? According to my understanding there should be one socket fd both for sending and receiving. Are you creating new socket fd for send part and closing recv fd? Please check and reply
You can also have two socket fds one for receiving and another for sending. RTP RFC-3550 doesnt say anything about implementation.
you should try to add media attribute in your invite, if you're using ulaw too then you can add:
a=rtpmap:0 PCMU/8000
Also try with a simpler test, instead of calling a softphone call:
301,1,Answer
301,2,Echo
Echo will capture the rtp stream from your client and send it back to you. If everything works fine then you can make a call between 2 working softphones and compare the traces with your client.
Also if possible try to post your dialplan and both users configuration.
(samll tip: if you enable canreinvite=yes or directrtpsetup=yes for both users they will be able to exchange rtp stream directly between each other instead of using asterisk as bridge)
Sounds like Asterisk may be attempting to re-INVITE your call so that it flows directly between your SIP app and the softphone. The advantage of Asterisk doing that is it makes the media path more efficient, the Asterisk server will no longer be bridging the call media only the signalling. The disadvantage is it can cause problems getting the RTP through if there are NATs involved or if a SIP user agent didn't support re-INVITEs, which may be the case with yours.
If it is a re-INVITE issue then firstly you should be able to see the extra INVITE request arrive at your SIP app or on the Asterisk console using a SIP debug. Secondly you can stop Asterisk doing re-INVITEs by setting canreinvite=no on the SIP account you are using.
For the record, I've decided to try a different PBX. I've downloaded and installed the 3CX Phone System and with this PBX everything works perfectly!
Now, the client for the beta version uses Patton at his site, so I just hope that this problem is specific to our Asterisk setup, and that it won't manifest there.
I've finally solved this problem! It turns out that the problem was not with the SIP messages, but with the code that set up the RTP session. I'm still not quite sure what went wrong, but it seems as though this code only works when the softphone is called directly (that is, not through a PBX) or when the softphone is on the same IP-address as the PBX.
This is the erroneous code:
public void startMedia(String peerIp,int peerPort,int receivePort,String format) throws IOException,MediaException,InvalidSessionAddressException
{
stopMedia();
this.format = format;
RTPSessionMgr rtpSessionMgr = new RTPSessionMgr();
rtpSessionMgr.initSession(new SessionAddress(),null,0.05,0.25);
InetAddress localhost = InetAddress.getLocalHost();
SessionAddress localAddr = new SessionAddress(localhost,receivePort,localhost,receivePort + 1);
InetAddress destAddr = InetAddress.getByName(peerIp);
rtpSessionMgr.startSession(localAddr,localAddr,new SessionAddress(destAddr,peerPort,destAddr,peerPort + 1),null);
sessionManager = rtpSessionMgr;
for (ReceiveStreamListener nextListener : receiveStreamListeners)
sessionManager.addReceiveStreamListener(nextListener);
}
This code was adapted from a book on SIP programming in Java (I guess that in order to preserve the author's reputation, I should not share which book that is).
When I went to look at the javadoc of the RTPManager class, I spotted some sample code in the documentation for setting up a unicast session and adapted it for my scenario. Here is the updated startMedia() method that works:
public void startMedia(String peerIp,int peerPort,int receivePort,String format,int sampleRate,int sampleSizeInBits) throws IOException,MediaException,InvalidSessionAddressException
{
stopMedia();
this.format = format;
this.sampleRate = sampleRate;
this.sampleSizeInBits = sampleSizeInBits;
sessionManager = RTPManager.newInstance();
SessionAddress localAddress = new SessionAddress(InetAddress.getLocalHost(),receivePort);
sessionManager.initialize(localAddress);
for (ReceiveStreamListener nextListener : receiveStreamListeners)
sessionManager.addReceiveStreamListener(nextListener);
InetAddress ipAddress = InetAddress.getByName(peerIp);
SessionAddress remoteAddress = new SessionAddress(ipAddress,peerPort);
sessionManager.addTarget(remoteAddress);
}
As you can see this code - although it uses the same classes - is quite different than that which I found in the book (which makes it hard to determine what the problem was), but it works perfectly!