Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
Hi Sven, I tried out your changes. I had to replace the $_SERVER['REMOTE_ADDR'] with Java's request.getRemoteAddr() since I'm using Jetty not Apache. I got the same results you got, which I also get using the something.invalid header. The peer connects from Chrome, I can dial my cellphone and make it ring, but the Chrome sipml5 client drops the call when the phone starts ringing. When I answer, the cellphone stays connected, but there is no audio. My suggestion is to post your changes to the user interface on the doubango Google Group as it will mean people don't need to modify the code to connect to Asterisk WS. https://groups.google.com/forum/?fromgroups=#!forum/doubango. See if they can incorporate your changes so we don't have to modify the library after each update. As far as the IP address goes, I'm not sure what this is doing since I still see the invalid domain in my SIP traces. James *I did some changes to the sipml5 client and wanted to share this with you guys... Actually only 2 simple changes...*https://github.com/mailsvb/sipml5 *- The main config section has been splitted and made a little more flexible, see *http://i45.tinypic.com/10x59o7.png - Main call.html file has been renamed to .php and some code has been added that will replace the something.invalid with the actual IP of your client PC. Currently I am able to register and at least make my softphone ring ;-) As soon as I answer the outgoing call from sipml5 in the softclient, I get an error in sipml5... You can find my console output here http://pastebin.com/jdkXSMSD I will continue investigating tomorrow... best regards, Sven -- James Mortensen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 8:20 PM, mailsvb mail...@gmail.com wrote: Hi, you need to build Asterisk with SRTP support... wget http://sourceforge.net/projects/srtp/files/latest/download -O srtp-latest.tgz tar -zxvf srtp-latest.tgz ./configure --prefix=/libsrtp make make install And for Asterisk... ./configure --with-srtp=/libsrtp this should work... Recompiled. Well... now at leat in ONE instance the signaling seems to behave correctly: when I dial from sipml5 to plain SIP. If the destination is sipml5, the destination browser goes into a funky state in which the live camera panel pops up but there doesn't seem to be a recognized ringing state. Here's the log from a sipml5-sipml5 call. The caller is 2010 and the callee is 2009. (12:40:06 is when I gave up and clicked hangup at the caller.) (Media? Heh, surely you jest.) [Aug 21 12:38:25] DEBUG[22872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: = Looking for Call ID: e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 (Checking From) --From tag wiwN3MEMrB3HGUmlel5V --To-tag [Aug 21 12:38:35] DEBUG[23469] logger.c: CALL_ID [C-0002] created by thread. [Aug 21 12:38:35] DEBUG[23469] acl.c: For destination '192.168.0.92', our source address is '192.168.0.111'. [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: Setting SIP_TRANSPORT_WS with address 192.168.0.111:5060 [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: Allocating new SIP dialog for e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 - INVITE (No RTP) [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: CALL_ID [C-0002] bound to thread. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Trying to put 'SIP/2.0 401' onto WS socket destined for 192.168.0.92:5060 [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: Call_ID [C-0002] being removed from thread. [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: = Looking for Call ID: e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 (Checking From) --From tag wiwN3MEMrB3HGUmlel5V --To-tag as39a7b995 [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: CALL_ID [C-0002] bound to thread. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Received ACK (6) - Command in SIP ACK [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Stopping retransmission on 'e4d7cda4-c4cb-932f-c084-ac6f87d27eb9' of Response 3106: Match Not Found [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: Call_ID [C-0002] being removed from thread. [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: = Looking for Call ID: e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 (Checking From) --From tag wiwN3MEMrB3HGUmlel5V --To-tag [Aug 21 12:38:35] DEBUG[23469] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: CALL_ID [C-0002] bound to thread. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb751d8dc' [Aug 21 12:38:35] DEBUG[23469][C-0002] res_rtp_asterisk.c: Allocated port 18704 for RTP instance '0xb751d8dc' [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] rtp_engine.c: RTP instance '0xb751d8dc' is setup and ready to go [Aug 21 12:38:35] DEBUG[23469][C-0002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb751d8dc' [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] VERBOSE[23469][C-0002] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Setting NAT on RTP to Off [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Processing session-level SDP o=- 1190078527 1 IN IP4 127.0.0.1... UNSUPPORTED OR FAILED. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Processing session-level SDP s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)... UNSUPPORTED OR FAILED.
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote: On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). ...which is in fact a 404 response from Asterisk. Here's the response I received: http://users.vialink.com.br/jcastro/refused.cap I suspect I am configuring something wrong, but what is it? Juan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. If this thread has taught me anything it's that there needs to be a complete wiki page, just copying/pasting what I'm saying here isn't enough. It's on my list. I won't call it a demo setup though... since it won't actually work yet. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote: On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). ...which is in fact a 404 response from Asterisk. Here's the response I received: http://users.vialink.com.br/jcastro/refused.cap I suspect I am configuring something wrong, but what is it? The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote: On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). ...which is in fact a 404 response from Asterisk. Here's the response I received: http://users.vialink.com.br/jcastro/refused.cap Well, of course unless you changed the port as you did in which case 80 in the URL instead of 8088. That is all! As I've said previously though, you won't get bidirectional audio or video flowing so trying that will fail, and it's known that it will fail. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp jc...@digium.com wrote: - Original Message - Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. If this thread has taught me anything it's that there needs to be a complete wiki page, just copying/pasting what I'm saying here isn't enough. It's on my list. I won't call it a demo setup though... since it won't actually work yet. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Agreed, but we need something and a place for comments. The wiki is great because we can rename and move things when they are no longer relevant to our needs. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 11:53 AM, Joshua Colp jc...@digium.com wrote:
- Original Message -
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br
wrote:
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro
jcas...@instant.com.br wrote:
I still get unauthorized from sipml5 with these modifications. I
used port 80 instead of 8088 (no other webserver listening on 80),
was
that wrong?
Correction. It's actually Failed to connect to the server. I set
the
proxy address and port correctly in sipml5's call.htm (it registers
on
Kamailio).
...which is in fact a 404 response from Asterisk. Here's the response
I received: http://users.vialink.com.br/jcastro/refused.cap
I suspect I am configuring something wrong, but what is it?
The complete URL to use is http://asterisk IP address or host:8088/ws
Note the /ws at the end. WebSocket support is only available there. Doing
otherwise would have required core HTTP server changes, which I wanted to
avoid. Depending on what you are testing with you may need to change it
slightly to add that in.
Well, I did the following changes in sipml5 and now I get a Bad
Request on REGISTER, instead of 404. Clearly, I'm still missing
something. Here are the changes I made:
Index: call.htm
===
--- call.htm(revision 68)
+++ call.htm(working copy)
@@ -351,8 +351,9 @@
// we will connect to one of them and let the
balancer to choose the right one (less connected sockets)
// each port can accept up to 65K connections which
means that the cloud can manage 325K active connections
// the number of port will be increased or decreased
based on the current trafic
-i_port = 4062 + (((new Date().getTime()) % 5) * 1000);
-s_proxy = sipml5.org;
+// i_port = 4062 + (((new Date().getTime()) % 5) * 1000);
+i_port = 80;
+s_proxy = 192.168.0.111;
}
// create a new SIP stack. Not mandatory as it's possible
to reuse the same satck
Index: src/tinySIP/src/tsip_stack.js
===
--- src/tinySIP/src/tsip_stack.js (revision 68)
+++ src/tinySIP/src/tsip_stack.js (working copy)
@@ -351,7 +351,7 @@
return -2;
}
-tsk_utils_log_info(SIP stack start: proxy=' +
this.network.s_proxy_cscf_host + : + this.network.i_proxy_cscf_port
+ ', realm=' + this.network.o_uri_realm + ', impi=' +
this.identity.s_impi + ', impu=' + this.identity.o_uri_impu + ');
+tsk_utils_log_info(SIP stack start: proxy=' +
this.network.s_proxy_cscf_host + : + this.network.i_proxy_cscf_port
+ /ws', realm=' + this.network.o_uri_realm + ', impi=' +
this.identity.s_impi + ', impu=' + this.identity.o_uri_impu + ');
this.network.o_transport =
this.o_layer_transport.transport_new(this.network.e_proxy_cscf_type,
this.network.s_proxy_cscf_host, this.network.i_proxy_cscf_port, SIP
Transport, __tsip_stack_transport_callback);
if (!this.network.o_transport) {
@@ -716,4 +716,4 @@
}
return 0;
-}
\ No newline at end of file
+}
Index: src/tinySIP/src/transports/tsip_transport.js
===
--- src/tinySIP/src/transports/tsip_transport.js(revision 68)
+++ src/tinySIP/src/transports/tsip_transport.js(working copy)
@@ -368,7 +368,7 @@
return -1;
}
-var s_url = tsk_string_format({0}://{1}:{2},o_self.s_protocol,
o_self.s_host, o_self.i_port);
+var s_url =
tsk_string_format({0}://{1}:{2}/ws,o_self.s_protocol, o_self.s_host,
o_self.i_port);
tsk_utils_log_info(Connecting to '+s_url+');
o_self.o_ws = new WebSocket(s_url, 'sip');
o_self.o_ws.binaryType = arraybuffer;
@@ -458,7 +458,7 @@
}
var b_isInternetExplorer = (WebRtc4all_GetType() == WebRtcType_e.IE);
-var s_url = tsk_string_format({0}://{1}:{2},o_self.s_protocol,
o_self.s_host, o_self.i_port);
+var s_url =
tsk_string_format({0}://{1}:{2}/ws,o_self.s_protocol, o_self.s_host,
o_self.i_port);
tsk_utils_log_info(Connecting to '+s_url+');
if(b_isInternetExplorer){
o_self.o_transport = new ActiveXObject(webrtc4ie.NetTransport);
@@ -480,7 +480,7 @@
if(o_self.o_transport.defaultDestAddr
o_self.o_transport.defaultDestPort){
o_self.s_host = o_self.o_transport.defaultDestAddr;
o_self.i_port = o_self.o_transport.defaultDestPort;
-tsk_utils_log_info(Transport default destination= +
o_self.s_host + : + o_self.i_port);
+tsk_utils_log_info(Transport default destination= +
o_self.s_host + : + o_self.i_port + /ws);
}
o_self.b_started = true;
o_self.signal(tsip_transport_event_type_e.STARTED, Network
transport started, null);
Index: src/tinyMEDIA/src/tmedia_session_jsep.js
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Juan Matt just opened https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document some of this. Feel free to pipe in. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
Put my sipml5 changes there. By the way, this is what happens when I try to call a X-Lite extension from a sipml5 extension: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:24:02] ERROR[22737][C-0009]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:24:02] WARNING[22737][C-0009]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Trying to do the reverse... X-Lite stays in Calling... - in sipml5, the right pane, with the local webcam thumbnailm, pops up, but no Answer button. Only Call and Hangup. Also, after a lng time, I get a ringing tone in X-Lite. And the webcam thing never goes away in sipml5. What I get in the log is just this: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 -- Executing [2010@demo:1] Dial(SIP/2012-0004, SIP/2010) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/2010 jcvmasterisk1*CLI sipml5 to sipml5: Not acceptable here. And the destination extension is totally inert. Log: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:30:58] ERROR[22747][C-000c]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:30:58] WARNING[22747][C-000c]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Meh, same thing as simpl5-to-plain-SIP. Juan On Mon, Aug 20, 2012 at 4:00 PM, Andrew Latham lath...@gmail.com wrote: On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support along with a work around via configuration. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Juan Matt just opened https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document some of this. Feel free to pipe in. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
Hi, you need to build Asterisk with SRTP support... *wget http://sourceforge.net/projects/srtp/files/latest/download -O srtp-latest.tgz tar -zxvf srtp-latest.tgz ./configure --prefix=/libsrtp make make install* *And for Asterisk...* *./configure --with-srtp=/libsrtp* * * *this should work...* * * *I did some changes to the sipml5 client and wanted to share this with you guys... Actually only 2 simple changes...* https://github.com/mailsvb/sipml5 *- The main config section has been splitted and made a little more flexible, see *http://i45.tinypic.com/10x59o7.png - Main call.html file has been renamed to .php and some code has been added that will replace the something.invalid with the actual IP of your client PC. Currently I am able to register and at least make my softphone ring ;-) As soon as I answer the outgoing call from sipml5 in the softclient, I get an error in sipml5... You can find my console output here http://pastebin.com/jdkXSMSD I will continue investigating tomorrow... best regards, Sven 2012/8/20 Juan Castro jcas...@instant.com.br Put my sipml5 changes there. By the way, this is what happens when I try to call a X-Lite extension from a sipml5 extension: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:24:02] ERROR[22737][C-0009]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:24:02] WARNING[22737][C-0009]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Trying to do the reverse... X-Lite stays in Calling... - in sipml5, the right pane, with the local webcam thumbnailm, pops up, but no Answer button. Only Call and Hangup. Also, after a lng time, I get a ringing tone in X-Lite. And the webcam thing never goes away in sipml5. What I get in the log is just this: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 -- Executing [2010@demo:1] Dial(SIP/2012-0004, SIP/2010) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/2010 jcvmasterisk1*CLI sipml5 to sipml5: Not acceptable here. And the destination extension is totally inert. Log: jcvmasterisk1*CLI == Using SIP RTP CoS mark 5 [Aug 20 17:30:58] ERROR[22747][C-000c]: chan_sip.c:32140 setup_srtp: No SRTP module loaded, can't setup SRTP session. [Aug 20 17:30:58] WARNING[22747][C-000c]: chan_sip.c:9974 process_sdp: Can't provide secure audio requested in SDP offer jcvmasterisk1*CLI Meh, same thing as simpl5-to-plain-SIP. Juan On Mon, Aug 20, 2012 at 4:00 PM, Andrew Latham lath...@gmail.com wrote: On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote: Hoo-hah. It registers. Progress! Now... media. Or not. On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote: - Original Message - The complete URL to use is http://asterisk IP address or host:8088/ws Note the /ws at the end. WebSocket support is only available there. Doing otherwise would have required core HTTP server changes, which I wanted to avoid. Depending on what you are testing with you may need to change it slightly to add that in. Well, I did the following changes in sipml5 and now I get a Bad Request on REGISTER, instead of 404. Clearly, I'm still missing something. Here are the changes I made: You are probably getting hit by a bug in Asterisk 11 that has been fixed. It's noted here in the wiki page I'm working on: https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Supportalong with a work around via configuration. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Juan Matt just opened https://issues.asterisk.org/jira/browse/ASTERISK-20267 to document some of this. Feel free to pipe in. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Juan Carlos Castro y
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Greetings, -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- Websocket support is being actively worked on. HTTP support should be enabled in manager.conf and http.conf first. --- manager.conf --- [general] enabled = yes webenabled = yes --- http.conf --- [general] enabled=yes bindaddr=0.0.0.0 bindport=8088 -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
- Original Message - On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Websocket support is being actively worked on. HTTP support should be enabled in manager.conf and http.conf first. Hola! The above will get the HTTP server portion going, but here's some other items: 1. transport=ws must be added to the peer/friend/user in sip.conf 2. avpf=yes must be set for that peer/friend/user as well. Depending on what you are testing with this can get you a little further. If you are using Chrome things will not quite work, yet. While they have made considerable progress becoming compliant with the ICE specification (SDP is now almost proper) it seems as though their STUN implementation is still not there yet. Completely valid packets sent by the library we use just seem to be ignored. Patience is a virtue really as things are still evolving. As well I will be working on a wiki page that will describe this stuff in detail. I was holding off until things were a bit more there but as people are at least trying it shall appear soon. Cheers, -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote: - Original Message - On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Websocket support is being actively worked on. HTTP support should be enabled in manager.conf and http.conf first. Hola! The above will get the HTTP server portion going, but here's some other items: 1. transport=ws must be added to the peer/friend/user in sip.conf 2. avpf=yes must be set for that peer/friend/user as well. Depending on what you are testing with this can get you a little further. If you are using Chrome things will not quite work, yet. While they have made considerable progress becoming compliant with the ICE specification (SDP is now almost proper) it seems as though their STUN implementation is still not there yet. Completely valid packets sent by the library we use just seem to be ignored. Patience is a virtue really as things are still evolving. As well I will be working on a wiki page that will describe this stuff in detail. I was holding off until things were a bit more there but as people are at least trying it shall appear soon. Cheers, -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- ___ Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 4:07 PM, Andrew Latham lath...@gmail.com wrote: On Fri, Aug 17, 2012 at 4:01 PM, Joshua Colp jc...@digium.com wrote: - Original Message - On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Websocket support is being actively worked on. HTTP support should be enabled in manager.conf and http.conf first. Hola! The above will get the HTTP server portion going, but here's some other items: 1. transport=ws must be added to the peer/friend/user in sip.conf 2. avpf=yes must be set for that peer/friend/user as well. Depending on what you are testing with this can get you a little further. If you are using Chrome things will not quite work, yet. While they have made considerable progress becoming compliant with the ICE specification (SDP is now almost proper) it seems as though their STUN implementation is still not there yet. Completely valid packets sent by the library we use just seem to be ignored. Patience is a virtue really as things are still evolving. As well I will be working on a wiki page that will describe this stuff in detail. I was holding off until things were a bit more there but as people are at least trying it shall appear soon. Cheers, -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- ___ Joshua Can you copy and past into a wiki page for everyone's benefit? Maybe https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or like page would be good. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ s/past/paste/ oops -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 5:01 PM, Joshua Colp jc...@digium.com wrote: - Original Message - On Fri, Aug 17, 2012 at 2:45 PM, Juan Castro jcas...@instant.com.br wrote: I see no indication of how to do this in sip.conf, and when I start Asterisk, it doesn't wait on port 80. Websocket support is being actively worked on. HTTP support should be enabled in manager.conf and http.conf first. Hola! The above will get the HTTP server portion going, but here's some other items: 1. transport=ws must be added to the peer/friend/user in sip.conf 2. avpf=yes must be set for that peer/friend/user as well. Depending on what you are testing with this can get you a little further. If you are using Chrome things will not quite work, yet. While they have made considerable progress becoming compliant with the ICE specification (SDP is now almost proper) it seems as though their STUN implementation is still not there yet. Completely valid packets sent by the library we use just seem to be ignored. Patience is a virtue really as things are still evolving. I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote: I still get unauthorized from sipml5 with these modifications. I used port 80 instead of 8088 (no other webserver listening on 80), was that wrong? Correction. It's actually Failed to connect to the server. I set the proxy address and port correctly in sipml5's call.htm (it registers on Kamailio). -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
