Problem with flex client connecting to mod_rtmp
Hello, today I tried the flex client with the mod_rtmp implementation. On a brandnew freeswitch I installed mod_rtmp and copied the flex direrectory to a web server and loaded the web page. However the flex client does not connect: Here's my freeswitch log 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session [f35a192c-0d02-4320-89e9-775a3573ee25] 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=342 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=12247364 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone', flex does connect. But in my freeswitch this fails. I have traced the network traffic and I can see that there is information flow between freeswitch and the client on connect request. Somebody has an idea where to look further? -- With kind regards Peter
Hi Peter, Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx the XXX would be the IP of your FreeSwitch Server. did you configure your rtmp.conf.xml and have a matching context in your dial plan? In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx"; jack On 8/6/2012 5:44 PM, Peter Steinbach wrote:
Hello,
today I tried the flex client with the mod_rtmp implementation. On a brandnew freeswitch I installed mod_rtmp and copied the flex direrectory to a web server and loaded the web page.
However the flex client does not connect: Here's my freeswitch log 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session [f35a192c-0d02-4320-89e9-775a3573ee25] 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=342 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=12247364 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone', flex does connect.
But in my freeswitch this fails. I have traced the network traffic and I can see that there is information flow between freeswitch and the client on connect request.
Somebody has an idea where to look further?
Hello Jack, here are some answers to your your questions:
Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server. I am still in the "connect" state, I am not logging in yet. The webpage tells me "Connecting... " and waits forever
did you configure your rtmp.conf.xml and have a matching context in your dial plan? I am still in the "connect" state
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL =
"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx"; I did not have a rtmpIP var, so I added it to the code, but no change.
In fact as I can see while grepping the network traffic, the Flex app and Freeswicth are negociating their protocol and freeswitch logs a "Sent connect reply", but this is very diferent from what is sent from the conference.freeswitch.org server. Best regards Peter On 08/07/12 06:26, Jack wrote:
Hi Peter, Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx the XXX would be the IP of your FreeSwitch Server.
did you configure your rtmp.conf.xml and have a matching context in your dial plan?
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
jack
On 8/6/2012 5:44 PM, Peter Steinbach wrote:
Hello,
today I tried the flex client with the mod_rtmp implementation. On a brandnew freeswitch I installed mod_rtmp and copied the flex direrectory to a web server and loaded the web page.
However the flex client does not connect: Here's my freeswitch log 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session [f35a192c-0d02-4320-89e9-775a3573ee25] 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=342 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=12247364 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone', flex does connect.
But in my freeswitch this fails. I have traced the network traffic and I can see that there is information flow between freeswitch and the client on connect request.
Somebody has an idea where to look further?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi, have you try putting: var flashvars = { rtmp_url: 'rtmp://xxx.xxx.xxx.xxx' }; var params = { allowScriptAccess: 'always' }; Also, you should remove the "phone" part from the url. Regards, Antonio On Tue, 2012-08-07 at 10:11 +0200, Peter Steinbach wrote:
Hello Jack,
here are some answers to your your questions:
Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server. I am still in the "connect" state, I am not logging in yet. The webpage tells me "Connecting... " and waits forever
did you configure your rtmp.conf.xml and have a matching context in your dial plan? I am still in the "connect" state
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL =
"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx"; I did not have a rtmpIP var, so I added it to the code, but no change.
In fact as I can see while grepping the network traffic, the Flex app and Freeswicth are negociating their protocol and freeswitch logs a "Sent connect reply", but this is very diferent from what is sent from the conference.freeswitch.org server.
Best regards Peter
On 08/07/12 06:26, Jack wrote:
Hi Peter, Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx the XXX would be the IP of your FreeSwitch Server.
did you configure your rtmp.conf.xml and have a matching context in your dial plan?
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
jack
On 8/6/2012 5:44 PM, Peter Steinbach wrote:
Hello,
today I tried the flex client with the mod_rtmp implementation. On a brandnew freeswitch I installed mod_rtmp and copied the flex direrectory to a web server and loaded the web page.
However the flex client does not connect: Here's my freeswitch log 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session [f35a192c-0d02-4320-89e9-775a3573ee25] 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=342 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=12247364 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone', flex does connect.
But in my freeswitch this fails. I have traced the network traffic and I can see that there is information flow between freeswitch and the client on connect request.
Somebody has an idea where to look further?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Un cordial saludo / Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com
Hello Antonio, I added this:
var params = { allowScriptAccess: 'always' };
and removed the /phone from the URL. This did not change anything. Log is still like this: 2012-08-07 11:56:48.105667 [NOTICE] mod_rtmp.c:744 New RTMP session [b31edb8a-50d3-4100-9b7f-ab49b6ba06b5] 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=331 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 11:56:48.205670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=11917881 stream_id=0x0] len=4 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:945 Set window size: 131072 bytes Ngrep on the traffic shows the following last packets starting with connect: (192.168.178.103 is the PC, 192.168.178.220 is Freeswitch) T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP] ..............................................................................................K........connect.?..........app.....flashVer...LNX 11,2,202,236..swfUrl..,http://my.domain.com/flex/freeswitch.swf..tcUrl...rtmp:/./my.domain.com..fpa............... ## T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] .... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] .... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ..... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ...... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ ## T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP] ...9............ Then traffic stops. The packet freeswitch answeres after the connect requests is (from wireshark): 2.109254 192.168.178.220 192.168.178.220 1935 192.168.178.103 36804 RTMP Unknown (0x7) | Unknown (0xf3) | Unknown (0x21) | Unknown (0x50) | Unknown (0xff)[Packet size limited during capture] With kind regards Peter
Hi Peter, Do you have some kind of firewall on your computer? did you try to disabled it? Regards, António On Tue, 2012-08-07 at 12:26 +0200, Peter Steinbach wrote:
Hello Antonio,
I added this:
var params = { allowScriptAccess: 'always' };
and removed the /phone from the URL. This did not change anything. Log is still like this:
2012-08-07 11:56:48.105667 [NOTICE] mod_rtmp.c:744 New RTMP session [b31edb8a-50d3-4100-9b7f-ab49b6ba06b5] 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=331 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 11:56:48.205670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=11917881 stream_id=0x0] len=4 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
Ngrep on the traffic shows the following last packets starting with connect: (192.168.178.103 is the PC, 192.168.178.220 is Freeswitch) T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP] ..............................................................................................K........connect.?..........app.....flashVer...LNX 11,2,202,236..swfUrl..,http://my.domain.com/flex/freeswitch.swf..tcUrl...rtmp:/./my.domain.com..fpa............... ## T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] .... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] .... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ..... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ...... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ ## T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP] ...9............ Then traffic stops.
The packet freeswitch answeres after the connect requests is (from wireshark): 2.109254 192.168.178.220 192.168.178.220 1935 192.168.178.103 36804 RTMP Unknown (0x7) | Unknown (0xf3) | Unknown (0x21) | Unknown (0x50) | Unknown (0xff)[Packet size limited during capture]
With kind regards Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Un cordial saludo / Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com
Hello Antonio, it's all on the local network and there is no firewall between. I can see that messages are exchanged in both directions on port 1935. But the right connect mesage seems not to be sent by freeswitch. Best regards Peter On 08/08/12 11:03, Antonio wrote:
Hi Peter,
Do you have some kind of firewall on your computer? did you try to disabled it?
Regards, António
On Tue, 2012-08-07 at 12:26 +0200, Peter Steinbach wrote:
Hello Antonio,
I added this:
var params = { allowScriptAccess: 'always' };
and removed the /phone from the URL. This did not change anything. Log is still like this:
2012-08-07 11:56:48.105667 [NOTICE] mod_rtmp.c:744 New RTMP session [b31edb8a-50d3-4100-9b7f-ab49b6ba06b5] 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=331 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 11:56:48.205670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 11:56:48.205670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=11917881 stream_id=0x0] len=4 2012-08-07 11:56:48.215667 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
Ngrep on the traffic shows the following last packets starting with connect: (192.168.178.103 is the PC, 192.168.178.220 is Freeswitch) T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP] ..............................................................................................K........connect.?..........app.....flashVer...LNX 11,2,202,236..swfUrl..,http://my.domain.com/flex/freeswitch.swf..tcUrl...rtmp:/./my.domain.com..fpa............... ## T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] .... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] .... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ..... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ...... # T 192.168.178.220:1935 -> 192.168.178.103:48179 [AP] ............ ## T 192.168.178.103:48179 -> 192.168.178.220:1935 [AP] ...9............ Then traffic stops.
The packet freeswitch answeres after the connect requests is (from wireshark): 2.109254 192.168.178.220 192.168.178.220 1935 192.168.178.103 36804 RTMP Unknown (0x7) | Unknown (0xf3) | Unknown (0x21) | Unknown (0x50) | Unknown (0xff)[Packet size limited during capture]
With kind regards Peter
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
--
Un cordial saludo / Best regards,
_________________________
António Silva
E-mail:asilva@wirelessmundi.com <mailto:asilva@wirelessmundi.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Peter, Here is my rtmp.conf.xml: <configuration name="rtmp.conf" description="RTMP Endpoint"> <profiles> <profile name="default"> <settings> <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" /> <!-- If this is set to true, no unauthenticated inbound calls will be allowed --> <param name="auth-calls" value="true" /> <!-- How much time should the clients buffer the media stream (miliseconds) --> <param name="buffer-len" value="50" /> <!-- Sets the maximum size of outbound RTMP chunks --> <param name="chunksize" value="512" /> </settings> </profile> </profiles> </configuration> It is important to specify the context and then make sure you have a condition in your corresponding dial plan that will catch your call. Jack On 8/7/2012 1:11 AM, Peter Steinbach wrote:
Hello Jack,
here are some answers to your your questions:
Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server. I am still in the "connect" state, I am not logging in yet. The webpage tells me "Connecting... " and waits forever
did you configure your rtmp.conf.xml and have a matching context in your dial plan? I am still in the "connect" state
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL =
"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx"; I did not have a rtmpIP var, so I added it to the code, but no change.
In fact as I can see while grepping the network traffic, the Flex app and Freeswicth are negociating their protocol and freeswitch logs a "Sent connect reply", but this is very diferent from what is sent from the conference.freeswitch.org server.
Best regards Peter
On 08/07/12 06:26, Jack wrote:
Hi Peter, Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx the XXX would be the IP of your FreeSwitch Server.
did you configure your rtmp.conf.xml and have a matching context in your dial plan?
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
jack
On 8/6/2012 5:44 PM, Peter Steinbach wrote:
Hello,
today I tried the flex client with the mod_rtmp implementation. On a brandnew freeswitch I installed mod_rtmp and copied the flex direrectory to a web server and loaded the web page.
However the flex client does not connect: Here's my freeswitch log 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session [f35a192c-0d02-4320-89e9-775a3573ee25] 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=342 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=12247364 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone', flex does connect.
But in my freeswitch this fails. I have traced the network traffic and I can see that there is information flow between freeswitch and the client on connect request.
Somebody has an idea where to look further?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks Jack, but as mentioned we are at an earlier stage. So far the Flex client has not yet conneted to Freeswitch. We still have a connect problem. Best regards Peter On 08/07/12 17:51, Jack wrote:
Peter, Here is my rtmp.conf.xml:
<configuration name="rtmp.conf" description="RTMP Endpoint"> <profiles> <profile name="default"> <settings> <param name="bind-address" value="0.0.0.0:1935" /> <param name="context" value="public" /> <param name="dialplan" value="XML" /> <!-- If this is set to true, no unauthenticated inbound calls will be allowed --> <param name="auth-calls" value="true" /> <!-- How much time should the clients buffer the media stream (miliseconds) --> <param name="buffer-len" value="50" /> <!-- Sets the maximum size of outbound RTMP chunks --> <param name="chunksize" value="512" /> </settings> </profile> </profiles> </configuration>
It is important to specify the context and then make sure you have a condition in your corresponding dial plan that will catch your call.
Jack
On 8/7/2012 1:11 AM, Peter Steinbach wrote:
Hello Jack,
here are some answers to your your questions:
Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx >the XXX would be the IP of your FreeSwitch Server. I am still in the "connect" state, I am not logging in yet. The webpage tells me "Connecting... " and waits forever
did you configure your rtmp.conf.xml and have a matching context in your dial plan? I am still in the "connect" state
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL =
"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx"; I did not have a rtmpIP var, so I added it to the code, but no change.
In fact as I can see while grepping the network traffic, the Flex app and Freeswicth are negociating their protocol and freeswitch logs a "Sent connect reply", but this is very diferent from what is sent from the conference.freeswitch.org server.
Best regards Peter
On 08/07/12 06:26, Jack wrote:
Hi Peter, Make sure you use a fully qualified username not just the extension 1001@xxx.xxx.xxx.xxx the XXX would be the IP of your FreeSwitch Server.
did you configure your rtmp.conf.xml and have a matching context in your dial plan?
In the web page make sure these two vars are set to YOUR Freeswitch server ip: var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
jack
On 8/6/2012 5:44 PM, Peter Steinbach wrote:
Hello,
today I tried the flex client with the mod_rtmp implementation. On a brandnew freeswitch I installed mod_rtmp and copied the flex direrectory to a web server and loaded the web page.
However the flex client does not connect: Here's my freeswitch log 2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session [f35a192c-0d02-4320-89e9-775a3573ee25] 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=342 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for connect 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5 ts=12247364 stream_id=0x0] len=4 2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes
If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone', flex does connect.
But in my freeswitch this fails. I have traced the network traffic and I can see that there is information flow between freeswitch and the client on connect request.
Somebody has an idea where to look further?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
Join Us At ClueCon - Aug 7-9, 2012
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
participants (3)
-
Antonio
-
Jack
-
Peter Steinbach