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More details at http://code.google.com/p/rtmplite and http://code.google.com/p/siprtmp -- Copyright (c) 2007-2009, Mamta Singh. Copyright (c) 2010-2011, Kundan Singh. All rights reserved. Copyright (c) 2011-2012, Intencity Cloud Technologies. All rights reserved. Copyright (c) 2011, Cumulus Python. No rights reserved. -- The main program is rtmp.py. Please see the embedded documentation in that file. Some parts of the documentation are copied here. Other modules such as amf, util and multitask are used from elsewhere and contain their respective copyright notices. -- The siprtmp module implements a SIP-RTMP gateway. Please see the google code project for details on demo instructions and support information. The project description is contained in siprtmp.py file itself. The command line of siprtmp is same as rtmp, hence just replce rtmp by siprtmp in the following example to start the gateway. -- The rtmpclient module implements a simple RTMP client. Please see the documentation in rtmpclient.py source file on how it works and how it can be used. -- The rtmfp module implements a simple and imcomplete RTMFP rendezvous server. Please see the documentation in rtmfp.py source file on how it works. -- Dependencies: Python 2.6 and Python 2.5 -- Typically an application can launch this server as follows: $ python rtmp.py -d The -d option enables debug trace so you know what is happening in the server. To know the command line options use the -h option: $ python rtmp.py -h A test client is available in testClient directory, and can be compiled using Flex Builder. Alternatively, you can use the SWF file to launch from testClient/bin-debug after starting the server. Once you have launched the client in the browser, you can connect to local host by clicking on 'connect' button. Then click on publish button to publish a stream. Open another browser with same URL and first connect then play the same stream name. If everything works fine you should be able to see the video from first browser to the second browser. Similar, in the first browser, if you check the record box before publishing, it will create a new FLV file for the recorded stream. You can close the publishing stream and play the recorded stream to see your recording. Note that due to initial delay in timestamp (in case publish was clicked much later than connect), your played video will start appearing after some initial delay. ---------------------- LICENSING ----------------------------------- This SOFTWARE consists of all the source code and documentation in the "rtmplite" project. The SOFTWARE is released under dual license, open source (GPL) or alternative license. The SOFTWARE may be used according to Free Software GNU General Public License (GPL) version 3, or (at your option) any later version. 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rtmplite's Issues
List of sockets for RTP/RTCP keeps growing with every call.
Reported by Franck <[email protected]> on Jan 28, 2011
Fixed in p2p-sip's svn r26 (version 1.10)
Hi Kundan,
didn't have any news from you.
hope you're well.
after experiment with rfc3550
I noticed that with p2p and siprtmp
udp socket session still stays open undiefinetly
and the list of socket is growing after every call.
so despite of my bad knowledge in python
i got the trick below:
def close(self):
if _debug: print 'cleaning up sockets'
if self.rtp: self.rtp.close(); self.rtp = None
if self.rtcp: self.rtcp.close(); self.rtcp = None
if self.app: self.app = None
self = None
once the call hangup siprtmp clean completely the ocket
and avoid that all udp port raange to be reached.
Take care
---
Hi Franck,
I think I found the problem.
It was a terrible oversight on my part in rfc3550.py.
I have fixed it in svn r26 as shown in
http://code.google.com/p/p2p-sip/source/list
I have also attached rfc3550.py which you can use to replace yours in
p2p-sip/src/std/rfc3550.py
Thanks a lot for helping with this bug!
Original issue reported on code.google.com by [email protected] on 7 Feb 2011 at 4:43
Handle multiple IP interfaces correctly
First reported by Joshua Shaffner <[email protected]> on Mar 5, 2011.
In summary, if the machine has multiple IP addresses, the siprtmp listens on
one IP address for RTMP but another for SIP. Fix this, so that if IP address is
supplied from command line, always use that.
--
I was able to register VideoIO with Freeswitch via siprtmp gateway but
there was no audio when I make a call. How do you suggest to
troubleshoot the issue?
Thank you,
Joshua
---
Hi,
Can you start siprtmp with -d option, and send me the program trace as
a text file. That can give some idea about what might be going wrong
in SIP/SDP negotiation. Problem could be in SDP codec negotiation or
in media port behind NAT/firewall.
Also, for next step, can you check if you can make call to your
freeswitch using X-lite (or some other SIP client) with only speex
enabled?
--
Any idea why sometimes it it will show 10.0.1.10 which is correct but
sometimes it show 10.211.55.2 (no idea where that came from)?
I had to register, unregister, restart servers, etc until it shows
10.0.1.10 for the contact before I could make a call
j
User: [email protected]
Contact: "1001 videoio" <sip:[email protected]:52948>
--
from rtmplite
rtmp-register scheme= sip aor= [email protected] login= 1001 passwd= ****
display= 1001 videoio
User created on ('10.211.55.2', 50486)
registering addr= "1001 videoio" <sip:[email protected]> port= 50486
createClient <UserAgent [email protected]>
sending[356] to ('10.0.1.10', 5060)
REGISTER sip:10.0.1.10 SIP/2.0
Content-Length: 0
Via: SIP/2.0/UDP 10.211.55.2:50486;rport;branch=z9hG4bK-RdvxJKS0AmOqI8HIv90aQ..
From: "1001 videoio" <sip:[email protected]>;tag=190882210157
Expires: 180
To: "1001 videoio" <sip:[email protected]>
Contact: <sip:[email protected]:50486>
CSeq: 1 REGISTER
Max-Forwards: 70
Call-ID: [email protected]
--
ah, interesting, check this out, it looks like self.sockaddr in
voip.py picks up ip from vnic0 even tho i dont use it. I use en1
(10.0.1.10). What do you suggest?
ifconfig
lo0: flags=8049<UP,LOOPBACK,RUNNING,MULTICAST> mtu 16384
inet6 ::1 prefixlen 128
inet6 fe80::1%lo0 prefixlen 64 scopeid 0x1
inet 127.0.0.1 netmask 0xff000000
gif0: flags=8010<POINTOPOINT,MULTICAST> mtu 1280
stf0: flags=0<> mtu 1280
en0: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 1500
ether c4:2c:03:16:b2:c1
media: autoselect
status: inactive
fw0: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 4078
lladdr 78:ca:39:ff:fe:32:16:12
media: autoselect <full-duplex>
status: inactive
en1: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 1500
ether c8:bc:c8:b8:fd:3d
inet 10.0.1.10 netmask 0xffffff00 broadcast 10.0.1.255
media: autoselect
status: active
vnic0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500
ether 00:1c:42:00:00:08
inet 10.211.55.2 netmask 0xffffff00 broadcast 10.211.55.255
inet6 fe80::21c:42ff:fe00:8%vnic0 prefixlen 64 scopeid 0x7
inet6 fec0:0:0:fea9::1 prefixlen 64
media: autoselect
status: active
vnic1: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500
ether 00:1c:42:00:00:09
inet 10.37.129.2 netmask 0xffffff00 broadcast 10.37.129.255
inet6 fe80::21c:42ff:fe00:9%vnic1 prefixlen 64 scopeid 0x8
inet6 fec0:0:0:feaa::1 prefixlen 64
media: autoselect
status: active
---
okay, I am not sure what makes it work this time..maybe after I make
"ip6" thingy auto instead of off in my network setting (OSX)
I did test using xlite with just speex and could hear playback (MOH)
but not with flash/speex. here is the trace
register and answering the call from FS which will then park the call
and play back music (for off hook agent)
---
got it working this time..did not publish/play when accepting call, my bad :)
Original issue reported on code.google.com by kundan10 on 10 Mar 2011 at 10:21
BYE not understood by freeswitch/x-lite
First reported by Madovsky <[email protected]> on Mar 26, 2010 and last on Apr
22, 2010.
The basic problem was that when using Freeswitch and Xlite, the reverse
direction BYE was not getting through from siprtmp. (see attached).
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 10:09
Attachments:
Create server startup script for Linux
Reported by Madovsky <[email protected]> on Jul 23, 2010.
hope you're well...
I made a lsb script to start your rtmplite app as
a linux daemon (joined) hope you'll like it
don't know yet very well python. I think I need
holidays for my eyes after 15 years of computer ;)
Take care
Franck
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 2:13
Attachments:
StreamID switches randomly and breaks existing streaming
First reported by Gabor Janak <[email protected]> on Jun 25, 2010.
> Hello,
>
> I'm experimenting with your rtmplite component. And notice a little
> Problem. Is this known ?
>
> I connect to a svn rtmlite server from a Flash Applet
> 1 netconnection.
> 2 NetStreams - server assigns streamId 1 and streamdid2
>
> now i start rtmpclient file.flv rtmp://url/app?id=test
>
> it call createstream, and server assigns streamId=1
>
> after some seconds a weired happen.
>
> the server thinks, it get an Message from the rtmpclient
> with streamId=2 .... Ough ?
> now an assert throws.. because server don't know this streamId in this
> Client Instance.
>
> I'm currently debugging the rtmp.py and try to add debug code
> but i could not find where the streamId changes...
>
> If the server is started, without connected the doublestream Applet ...
> the error doesn't happen.
>
> If the double-stream client is disconnected... the error doesn't happen.
>
>
> Can you give me a Hint where to search ?
>
>
> Thanks in advantage...
>
> Bye
> Gabor
---
Hello again,
i found my Failure. I used a flv file which i recorded with the rtmpClient.
I don't know why,but at the beginning of the file a TextData Message is
written.
hexdump a.flv -C -n 100
00000000 46 4c 56 01 05 00 00 00 09 00 00 00 00 09 00 00
|FLV.............|
00000010 1d 00 09 cd 00 00 00 00 02 00 0d 73 74 61 74 75
|...........statu|
00000020 73 4d 65 73 73 61 67 65 02 00 0a 57 65 6c 63 6f
|sMessage...Welco|
00000030 6d 65 2e 2e 2e 00 00 00 28 09 00 07 ca 00 13 9a
|me......(.......|
00000040 00 00 00 00 12 00 00 84 02 82 a6 20 20 20 21 ff |...........
!.|
00000050 ff 31 01 01 01 0f ff f9 88 08 08 08 7f ff cc 40
|.1.............@|
00000060 40 40 43 ff |@@C.|
00000064
sended by NetStream.send("statusMessage","Welcome...");
it seems the rtmpclient wrote this Message into the FLV,
and now the old streamID 0x02 is inserted into the new Stream...
I will patch the rtmpClient, that this type of messages arn't written
into the flv :) or is this behavior planned ?
Bye
Gabor
---
> Hello,
>
> sorry, but this is the Last Email.
>
> I send you a small fix.
>
> the Problem is, that you cache the header in lastReadHeader
> by reference. and this Reference is in the message,too.
> if this Message is send over a stream, the stream send function
> calls msg.streamId=self.id (streamId)
> this changes the streamId of the cached Header ...
> and if the next message is not type0 it use the wrong streamId
>
> solution:
> @@ -523,7 +530,13 @@
> '''Method to send a Message or Command on this stream.'''
> if isinstance(msg, Command):
> msg = msg.toMessage()
> - msg.streamId = self.id
> + if msg.streamId != self.id:
> + #make a copy of the message, changing the orginal +
> #messages is BAD, because it can hold
> + #a reference to a cached header
> + old_msg=msg
> + msg=copy.deepcopy(old_msg)
> + msg.streamId = self.id
> # if _debug: print self,'send'
> if self.client is not None: self.client.writeMessage(msg)
>
> Attached a patch...
>
> I hope this helps.
>
> Thanks.
>
---
Hey,
Thanks a lot for identifying the problem. We will add your suggestion
to the SVN code soon.
Original issue reported on code.google.com by [email protected] on 4 Feb 2011 at 11:35
Integrate G.711 and H.264 available with Flash Player 11
Flash Player 11 (in beta) has new audio and video codecs support from
microphone and camera captured data. Integrated this with siprtmp/VideoPhone
and Flash-VideoIO.
Original issue reported on code.google.com by kundan10 on 27 Jul 2011 at 4:53
Exception in rtmpclient when connection is closed by remote
Reported by Jonas Borjesson on Apr 6, 2011.
File "rtmp.py",
line 283, in parseMessages
count = min(header.size - (len(data)), self.readChunkSize) # how much more
TypeError: unsupported operand type(s) for -: 'NoneType' and 'int'
Original issue reported on code.google.com by kundan10 on 16 Apr 2011 at 1:49
RTP sockets/ports are not closed in siprtmp
Reported by Hakan Kocaman <[email protected]> on May 12, 2010
Hi,
we are using your sip<>rtmp-gateway in a prototype together with freeswitch.
We are very pleased with it, but now we are seeing that there are a
lot of open ports :
Netstat -punta
[..]
udp 0 0 0.0.0.0:43595 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:25675 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:57803 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:19275 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:51148 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:51149 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:12494 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:38606 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:12495 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:38607 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:56016 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:62416 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:56017 0.0.0.0:*
10076/python
udp 0 0 0.0.0.0:62417 0.0.0.0:*
10076/python
[..]
There are at the moment 166 open ports, although there is no activity
from our flash-client.
Is this expected behavior ? Is their something like a garbagecollector
involved ?
I must admit, that there is no python-specialist in our team for this project,
so we don't know is this is OK.
We would appreciate any help we can get from your site.
Best regards from germany
Freundliche Grรผรe
Hakan Kocaman
Software-Development
---
Hi Hakan,
Thanks for trying out siprtmp.
These port numbers are related to RTP/RTCP socket that are opened for
media stream.
I have a close() method in the voip.py::MediaSession and
rfc3550::RTPNetwork classes. The close
method of RTPNetwork gets called from MediaSession.close(), but I
guess the siprtmp is not calling the close() method.
I will try to look at the siprtmp.py's cleanup code to make sure that
it correctly closes the SIP and RTP sessions.
---
Thank you for your swift reply!
Thanks a lot for having a look at it.
Indeed, I believe that we suffer from unclean session aborts here.
The software we're using siprtmp against is freeswitch which uses
nokia's sofia sip implementation. If you feel that the aborted sessions
are owed to a bug in freeswitch or sofia sip, let me know and I'll hook
you up with the right person for the problem over at the freeswitch
project.
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 10:51
ffmpeg encoder connection: KeyError: 'objectEncoding'
On Ubuntu 10.10 I download rtmplite-6.0. I start the application...
python rtmp.py -d
On the same machine, I attempt to connect to it with a live rtmp stream from
ffmpeg....
ffmpeg -i "/somevideo.vob" -re -vcodec libx264 -vpre default -b 500000 -s
320x180 -strict experimental -g 25 -me_method zero -acodec aac -ab 96000 -ar
48000 -ac 2 -b 500K -f flv rtmp://127.0.0.1/live/livestream
I get the following output from rtmp.py...
connection received from ('127.0.0.1', 34624)
socket.read[23] calling recv()
socket.read[1537]
"\x03\x00\x19\x160\x00\x00\x00\x00gE\x8bk\xc6#{2i\x98<dsH3fQ\xdc\xb0t\xff\\I\x19
J\x94\xe8*\xecXUb)\x1f\x8e#\xcd|\xe8F\xbaX\x1b=\xab\xd7~P\xf2A\xb1.\xfb\x1e\xb7A
\xe3\xa9\xe2yF\xe1Eu|\x00_Q\xc2b\xd0[T\x08
\x12\xf8'\xb1M\x1b#\x16\x02\xe8\xe9\x16\x1f\xe7\xcd\x90...(1537)"
socket.write[1537]
"\x03\x00\x19\x160\x00\x00\x00\x00gE\x8bk\xc6#{2i\x98<dsH3fQ\xdc\xb0t\xff\\I\x19
J\x94\xe8*\xecXUb)\x1f\x8e#\xcd|\xe8F\xbaX\x1b=\xab\xd7~P\xf2A\xb1.\xfb\x1e\xb7A
\xe3\xa9\xe2yF\xe1Eu|\x00_Q\xc2b\xd0[T\x08
\x12\xf8'\xb1M\x1b#\x16\x02\xe8\xe9\x16\x1f\xe7\xcd\x90...(1537)"
socket.read[1536] calling recv()
socket.read[1536]
"\x00\x19\x160\x00\x00\x00\x00gE\x8bk\xc6#{2i\x98<dsH3fQ\xdc\xb0t\xff\\I\x19J\x9
4\xe8*\xecXUb)\x1f\x8e#\xcd|\xe8F\xbaX\x1b=\xab\xd7~P\xf2A\xb1.\xfb\x1e\xb7A\xe3
\xa9\xe2yF\xe1Eu|\x00_Q\xc2b\xd0[T\x08
\x12\xf8'\xb1M\x1b#\x16\x02\xe8\xe9\x16\x1f\xe7\xcd\x90\x11...(1536)"
socket.write[1536]
"\x00\x19\x160\x00\x00\x00\x00gE\x8bk\xc6#{2i\x98<dsH3fQ\xdc\xb0t\xff\\I\x19J\x9
4\xe8*\xecXUb)\x1f\x8e#\xcd|\xe8F\xbaX\x1b=\xab\xd7~P\xf2A\xb1.\xfb\x1e\xb7A\xe3
\xa9\xe2yF\xe1Eu|\x00_Q\xc2b\xd0[T\x08
\x12\xf8'\xb1M\x1b#\x16\x02\xe8\xe9\x16\x1f\xe7\xcd\x90\x11...(1536)"
socket.read[1] calling recv()
socket.read[97]
'\x03\x00\x00\x00\x00\x00U\x14\x00\x00\x00\x00\x02\x00\x07connect\x00?\xf0\x00\x
00\x00\x00\x00\x00\x03\x00\x03app\x02\x00\x04live\x00\x04type\x02\x00\nnonprivat
e\x00\x05tcUrl\x02\x00\x15rtmp://127.0.0.1/live\x00\x00\t'
Protocol.parseMessage msg= <Message header=<Header channel=3 time=0 size=85
type=20 streamId=0>
data='\x02\x00\x07connect\x00?\xf0\x00\x00\x00\x00\x00\x00\x03\x00\x03app\x02\x0
0\x04live\x00\x04type\x02\x00\nnonprivate\x00\x05tcUrl\x02\x00\x15rtmp://127.0.0
.1/live\x00\x00\t'>
Traceback (most recent call last):
File "rtmp.py", line 295, in parseMessages
yield self.messageReceived(msg)
File "/home/simonb/Documents/Source/rtmplite/multitask.py", line 963, in run_next
output = task.send(input)
File "/home/simonb/Documents/Source/rtmplite/multitask.py", line 182, in send
return self.task.send(value)
File "rtmp.py", line 611, in messageReceived
self.objectEncoding = self.agent['objectEncoding']
KeyError: 'objectEncoding'
Protocol.parseMessages exception None
socket.read[1] calling recv()
Original issue reported on code.google.com by [email protected] on 4 Feb 2011 at 6:39
CPU charge question
- siprtmp takes without calls about 2% of CPU and with one call about 10% with
a nocona 2 x Xeon
This is a little weird since there is no audio/video encoding.
is this the effect of RTMP to RTP loop ?
Original issue reported on code.google.com by [email protected] on 8 Feb 2011 at 11:08
Recorded FLV cannot be played by flowplayer or JW player
Reported by John K <[email protected]> on May 1, 2010
Hello! First, thanks for providing this. A lightweight RTMP server is
what I'm looking for. I'm trying to stream FLV videos. With the
supplied testclient, I can watch an FLV, but when I try to view it
using a commercial player like JW player or flowplayer, it just
displays animation that indicates it's loading but the video never
plays. For instance,
http://www.push-a-button.com/products/youglevista/rtmp.php is a site
that lets you play an rtmp stream you specify with flowplayer. About
the only reason I can think of at this point is that with my web host,
I'm using a non-standard port for my server, i.e. it's not 1935. If
you have any suggestions on how to make this work, I would greatly
appreciate it. If you don't have time to do support, I understand, but
if I can get this to work, I will definitely spread the word about
rtmplite at my web host support forum & also in JW player forum.
thanks,
John K
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 10:17
Memory leak in siprtmp and rfc3261.py
Reported by ลukasz Paszke <[email protected]> May 6, 2010
Hi,
Thank you for p2p-sip.
I am debugging memory leaks in siprtmp.py from rtmplite, and sipd.py from
p2p-sip. siprtmp.py and sipd.py are based on p2p-sip. I know that in sip.py
memory usage increases after every register request generated by asterisk
1.4.22 on reregistration. This is because ServerTransaction and Proxy objects
are not closed (removed from memory) after each successful register
transaction. Therefore there is a buildup in the number of this object over
time (40 MB over 4 days).
Can you advise what can be wrong and where to look for the solution in the code?
Best Regards / Pozdrawiam
ลukasz
---
I can try looking at the memory problem in rfc3261.py file.
There is a circular reference among transaction and Dialog/Proxy
object I feel. I think I had implemented a close() method in both the
classes. But may be the close() method is not getting invoked for
cleanup on the Proxy object.
Which tool are you using to detect the memory leak?
We can probably chat further on gtalk ([email protected]) and nail
this down....
---
I use pdb and objgraph from
http://www.lshift.net/blog/2008/11/14/tracing-python-memory-leaks
I am new to python (I have experience in ruby, a little bit like python).
I think the problem occurs because of typo in rfc3261.py. Please see
rfc3261.py.patch for more details. After this correction objects are correctly
destroyed during registers/reregisters with asterisk.
I found some other leak occurring every time siprtmp.py registers/reregisters
to sipd.py (increase in application memory consumption), but unfortunately I
cant find any application objects in memory space after application termination.
siprtmp.py also have a number of more complicated memory leaks.
---
Index: rfc3261.py
===================================================================
--- rfc3261.py (revision 917)
+++ rfc3261.py (revision 918)
@@ -560,7 +560,7 @@
def state():
def fset(self, value):
self._state = value
- if self._state == 'terminating': self.close() # automatically
close when state goes terminating
+ if self._state == 'terminated': self.close() # automatically close
when state goes terminating
def fget(self): return self._state
return locals()
state = property(**state())
---
Hi Lukasz,
Thanks for the patch. That was a tricky typo and thanks for detecting.
We will add the memory leak detection as our things to do. I agree
that siprtmp.py is pretty primitive -- it was a student project and
haven't been regressed for robustness.
Regards
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 11:21
AttributeError ClientTransaction object has no receivedRequest in siprtmp
Reported by bernard roubaud <[email protected]>on Sep 14, 2010.
Hi,
I am playing with siprtmp (6.0) and X-Lite (3.0 build 56125) as described i=
n the =93Testing=94 paragraph and I got the following error : AttributeErro=
r: 'ClientTransaction' object has no attribute 'receivedRequest'
Does I miss something in the configuration ?
Thanks
Bernard
C:\p2p-sip\src\app>python.exe sipd.py -d
starting agent on ('0.0.0.0', 5060) with transports ('udp',)
('10.184.83.33', 5062)=3D>('0.0.0.0', 5060) on type=3D2
REGISTER sip:10.184.83.33 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062
;branch=3Dz9hG4bK-d8754z-49044a660b56871e-1---d875
z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5062;rinstance=3D6d89135e21df93da>
To: "bernard"<sip:[email protected]>
From: "bernard"<sip:[email protected] <sip%[email protected]>
>;tag=3D57755873
Call-ID: MTE4ZTk3Mzc3YTIxMWI1OTIzNjRkZTk0ODdhYzk1ZTc.
CSeq: 1 REGISTER
Expires: 0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE=
,
IN
O
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
received request from stack REGISTER
save {}
locate sip:[email protected] {}
('0.0.0.0', 5060)=3D>('10.184.83.33', 5062) on type=3D2
SIP/2.0 200 OK
Content-Length: 0
Via: SIP/2.0/UDP 127.0.0.1:5062
;received=3D10.184.83.33;branch=3Dz9hG4bK-d8754z-490
4a660b56871e-1---d8754z-;rport=3D5062
From: "bernard" <sip:[email protected] <sip%[email protected]>
>;tag=3D57755873
Expires: 0
To: "bernard" <sip:[email protected] <sip%[email protected]>
>;tag=3D28744989463
CSeq: 1 REGISTER
Call-ID: MTE4ZTk3Mzc3YTIxMWI1OTIzNjRkZTk0ODdhYzk1ZTc.
('10.184.83.33', 5062)=3D>('0.0.0.0', 5060) on type=3D2
SUBSCRIBE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5062
;branch=3Dz9hG4bK-d8754z-a67c6e0157793427-1---d875
z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5062>
To: "bernard"<sip:[email protected]>
From: "bernard"<sip:[email protected] <sip%[email protected]>
>;tag=3D2607fb33
Call-ID: ZTJjZDUyNjBhYjM2Njc3MjI4YTk0NjQyMjgwZGUzYTk.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE=
,
IN
O
User-Agent: X-Lite release 1104o stamp 56125
Event: message-summary
Content-Length: 0
received request from stack SUBSCRIBE
proxying non-invite non-local request
sending request on stack SUBSCRIBE
('0.0.0.0', 5060)=3D>('10.184.83.33', 5060) on type=3D2
SUBSCRIBE sip:[email protected] SIP/2.0
Event: message-summary
Content-Length: 0
Via: SIP/2.0/UDP 10.184.83.33:5060
;rport;branch=3Dz9hG4bKlPGUl9DxKlheAqvOZPNE5A..
Via: SIP/2.0/UDP 127.0.0.1:5062
;received=3D10.184.83.33;rport=3D5062;branch=3Dz9hG4bK
d8754z-a67c6e0157793427-1---d8754z-
From: "bernard" <sip:[email protected] <sip%[email protected]>
>;tag=3D2607fb33
Expires: 300
To: "bernard" <sip:[email protected] <sip%[email protected]>>
Contact: <sip:[email protected]:5062>
CSeq: 1 SUBSCRIBE
Max-Forwards: 69
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: OPTIONS
Allow: BYE
Allow: REFER
Allow: NOTIFY
Allow: MESSAGE
Allow: SUBSCRIBE
Allow: INFO
Call-ID: ZTJjZDUyNjBhYjM2Njc3MjI4YTk0NjQyMjgwZGUzYTk.
User-Agent: X-Lite release 1104o stamp 56125
('10.184.83.33', 5060)=3D>('0.0.0.0', 5060) on type=3D2
SUBSCRIBE sip:[email protected] SIP/2.0
Event: message-summary
Content-Length: 0
Via: SIP/2.0/UDP 10.184.83.33:5060
;rport;branch=3Dz9hG4bKlPGUl9DxKlheAqvOZPNE5A..
Via: SIP/2.0/UDP 127.0.0.1:5062
;received=3D10.184.83.33;rport=3D5062;branch=3Dz9hG4bK
d8754z-a67c6e0157793427-1---d8754z-
From: "bernard" <sip:[email protected] <sip%[email protected]>
>;tag=3D2607fb33
Expires: 300
To: "bernard" <sip:[email protected] <sip%[email protected]>>
Contact: <sip:[email protected]:5062>
CSeq: 1 SUBSCRIBE
Max-Forwards: 69
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: OPTIONS
Allow: BYE
Allow: REFER
Allow: NOTIFY
Allow: MESSAGE
Allow: SUBSCRIBE
Allow: INFO
Call-ID: ZTJjZDUyNjBhYjM2Njc3MjI4YTk0NjQyMjgwZGUzYTk.
User-Agent: X-Lite release 1104o stamp 56125
Traceback (most recent call last):
File "sipd.py", line 110, in <module>
sipapi.run() # the loop to process the SIP listening point
File "c:\p2p-sip\src\app\sipapi.py", line 246, in run
try: multitask.run()
File "c:\p2p-sip\src\external\multitask.py", line 1196, in run
get_default_task_manager().run()
File "c:\p2p-sip\src\external\multitask.py", line 897, in run
self.run_next()
File "c:\p2p-sip\src\external\multitask.py", line 963, in run_next
output =3D task.send(input)
File "c:\p2p-sip\src\app\sipapi.py", line 151, in _sipreceiver
if data: stack.received(data, remote)
File "c:\p2p-sip\src\std\rfc3261.py", line 425, in received
self._receivedRequest(m, uri)
File "c:\p2p-sip\src\std\rfc3261.py", line 494, in _receivedRequest
t.receivedRequest(r)
AttributeError: 'ClientTransaction' object has no attribute
'receivedRequest'
C:\p2p-sip\src\app>
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 3:35
siprtmp_gevent.py fail on video transmit
Transmitting video fail (client closes connection) when running server by
siprtmp_gevent.py.
Problem in function FlashCliend.send(), there is no check for all data sent.
Solution: replace self.sock.send(data) to self.sock.sendall(data), or process
sending in loop with counting sent bytes.
Original issue reported on code.google.com by [email protected] on 28 May 2011 at 5:25
rtmpclient_gevent
Please help to build a rtmpclient_gevent.
Mean replace multitask by gevent in rtmpclient, so we can use with
siprtmp_gevent.py
Original issue reported on code.google.com by [email protected] on 27 Apr 2011 at 6:52
How to set User-Agent header?
- how to define "Agent" name ? for now the b2bua proxy shows "unknown" as User
agent (why not Kundan Sip ? :))
Original issue reported on code.google.com by [email protected] on 8 Feb 2011 at 11:06
Display name in UTF8
- UTF8 in display name is not supported, also the "*" in fromt of a number.
maybe it should be encoded in base64 also ?)
Original issue reported on code.google.com by [email protected] on 8 Feb 2011 at 11:07
exeception error in incoming calls with gevent version
Hi,
below the error trace when the ua receives a call.
it's not fatal but sometimes no audio
receive RTP exception (<class 'socket.error'>, error(9, 'Bad f
ile descriptor'), <traceback object at 0x2374878>)
Traceback (most recent call last):
File "/home/adobe/rtmplite/siprtmp.py", line 642, in receiveRTP
fd = sock.fileno()
File "<string>", line 1, in fileno
File "/usr/lib64/python2.5/site-packages/gevent-0.14.0dev-py2.5-linux-x86_64.e gg/gevent/socket.py", line 266, in _dummy
raise error(EBADF, 'Bad file descriptor')
error: (9, 'Bad file descriptor')
receive RTCP exception (<class 'socket.error'>, error(9, 'Bad file
descriptor'),
<traceback object at 0x23747e8>)
Exception greenlet.GreenletExit: GreenletExit() in <bound method
Network.__del__
of <__main__.Network object at
0x2375110>> ignored
receive RTP exception (<class 'socket.error'>, error(9, 'Bad file descriptor'),
<traceback object at 0x24253f8>)
Traceback (most recent call last):
File "/home/adobe/rtmplite/siprtmp.py", line 642, in receiveRTP
fd = sock.fileno()
File "<string>", line 1, in fileno
File "/usr/lib64/python2.5/site-packages/gevent-0.14.0dev-py2.5-linux-x86_64.e gg/gevent/socket.py", line 266, in _dummy
raise error(EBADF, 'Bad file descriptor')
error: (9, 'Bad file descriptor')
receive RTCP exception (<class 'socket.error'>, error(9, 'Bad file
descriptor'),
<traceback object at 0x24253f8>)
Exception greenlet.GreenletExit: GreenletExit() in <bound method
Network.__del__
of <__main__.Network object at
0x2422290>> ignored
I tried to figure it out without success.
seems that maybe the sock.fileno() is not ready at the time to send packet?
thanks
Original issue reported on code.google.com by [email protected] on 28 May 2011 at 7:52
Implement and test AMF3 support in rtmplite
AMF3 is now implemented in amf.py, but needs testing and integration with
rtmp.py.
Original issue reported on code.google.com by [email protected] on 14 Feb 2011 at 6:29
Server side playlist for radio station
Reported by daslicht <[email protected]> on Nov 24, 2010.
Hi would it be possible to built a server side playlist so that I
could built a radio station ?
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 4:26
Interoperate with Asterisk for Audio/Video
Reported by Borja SIXTO <[email protected]> on May 14, 2010
Hi,
I am Borja SIXTO from i6net.
We made working your Application (SipRTMP) with an Asterisk (audio and
video : using a transcoder in the Asterisk).
Are you interested by the changes done ?
Regards,
Borja
---
> Hi Borja,
>
> Thanks for the offer to send the changes.
> It will help a lot to the project if you can send your changes and any
> other lessons learned.
>
> Thanks.
---
Nice to get your response.
I do two modifications :
1/ enable Speex to be able to pass full audio calls with the Asterisk
(with Asterisk 1.6 it works fine, with a warning but the audio is ok).
2/ I had developp a video transcoder for the video h263sorenon
(integrated in the Asterisk 1.6), now I can call sip phone (h263, h263
and h264) and 3G video phones. For the video, the GW need to use 2
different netstreams to get a smooth audio/video.
It will be great to get the DTMF, but I don't know how to add the in
the SIP/sdp and how to send them from the P2P-sip API.
This is a prototype to make our native RTMP channel in the Asterisk.
Regards,
Borja
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 11:55
Attachments:
Python list to AMF in rtmplite
Reported by alfonz77 <[email protected]> on Nov 29, 2010.
Hello,
First of all congrats for the project I really like it simplicity. I
am trying to build a multi-player game with Flex and Python - I am a
beginner with Python. I noticed that Python lists are nor sent
correctly to Flash - actually when I try to send a list I get corrupt
data on the Flash side. Currently I make a workaround by converting
lists to dictionaries before sending. Am I missing something?
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 5:25
Add video transcoding between Flash Player and H.264 endpoint
Seems like Video is in huge demand...
Need to evaluate what it takes and whether it is feasible in realtime to
transcode between Sorenson and H.264.
Original issue reported on code.google.com by kundan10 on 10 Mar 2011 at 10:08
- Merged into: #49
No video in siprtmp using videophone.swf.
Reported by Rus <[email protected]> on May 9, 2010
Hello,
First thanks for this interesting project. Tried to make it work with
latest rtmplite and p2p-sip svn. The clients can connect to servers,
can view the initial camera video, can do the call, but just when the
call are
answered all cameras led became inactive and no video (only vertical rainbow
color bars) is shown. Logs are attached. I have the simple setup of
two boxes - first (10.1.0.100 is running siprtmp and sipd and
videophone.swf client named Dao, second (10.1.0.7) is running
videophone.swf client named acer. This is internal home network - no
firewalls and gateways - only hub connection. I see that contact field
in sipd logs are wrong for second client (it needs to be 10.1.0.7 but
seems like it is the same as for first client). Settings for
10.1.0.100 swf client are :
rtmp url - rtmp://localhost/sip
sip url - [email protected]
auth name - dao
auth pass - 1234 (do not care)
name - Dao
Settings for 10.1.0.7 swf client are :
rtmp url - rtmp://10.1.0.100/sip
sip url - [email protected]
auth name - acer
auth pass - 1234 (do not care)
name - Acer
What can be the issue ? Sipd.py and siprtmp.py logs are attached. Can
supply any additional info.
Thanks in advance.
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 10:25
Attachments:
siprtmp hanging on Python 2.7
First reported by James [email protected] on Mar 9, 2011.
--
Hi,
I've downloaded and setup siprtmp on a centos server. The default python
installation is 2.4 so i've done an alternate installation of 2.7. When i try
to run the server using python2.7 siprtmp.py -d it shows 'Flash Server Starts -
0.0.0.0:1935' and nothing happens after that. When i check on the ports using
netstat there's no service running on port 1935. Is there something am doing
wrong? How do i know when siprtmp is running?
james
--
Original issue reported on code.google.com by kundan10 on 10 Mar 2011 at 9:33
Problem with NetStream.bufferTime in rtmplite
Reported by Alon Gubkin <[email protected]> on Dec 1, 2010.
Are there any known issues with NetStream.bufferTime and rtmplite? For
some reason it doesn't work for me.
Thanks.
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 5:36
Working with Flash Media Encoder
Reported by Tomas Asmenybe <[email protected]> on Sep 2, 2010
Hello,
I tried to use flash media encoder 2.5 with your rtmp server, but it
throws error "keyerror: objectencoding"... Is it possible to use fme
with your server or should I look for something else? Any help would
be apreciated.
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 2:20
Configuration/method to set IP address in 'c' line of SDP
Use case:
When running on amazon cloud (or behind a NAT), listening address and
advertised address (in SDP) are same. Need a procedure or configuration to
specify the advertised address in SDP.
Original issue reported on code.google.com by [email protected] on 29 Mar 2011 at 12:31
AsyncError was unable to invoke onMetaData
Reported by Erald (gmx) <[email protected]> on Jun 3, 2010.
Hello,
This looks like the server I am looking for. Red5 and others are
complex and since I am really a noob on flash very difficult to
understand.
I understand it supports live streaming to which you can connect. This
I might want to use in the future. However I am interested in using
the server as a .flv streaming server using the jw player.
However I tried you example application to stream some files but get
always the error: streamErrorHandler() asyncError Error #2095:
flash.net.NetStream was unable to invoke callback onMetaData.
How should I avoid this and can I use this server as a rtmp .flv
streaming server.
Some guidance would be very nice on how to configure this.
Thank you for your help
Erald
Original issue reported on code.google.com by [email protected] on 4 Feb 2011 at 11:16
Easy Application override
Hi,
I tried rtmplite with openlaszlo, it works like a charm ! Only one thing. It's
not nice to have to recreate FlashServer to override application methods.
So, I did a patch to allow user to override Application class and to pass
"dict" to FlashServer constructor.
So you can do;
class MyApp(Application):
pass
class MyOtherApp(Application):
pass
And send it to constructor, for example:
app = FlashServer(dict({'url1': MyApp, 'url2'; MyOtherApp}))
Original issue reported on code.google.com by Metal3d on 19 Feb 2011 at 1:42
Attachments:
Close reader task and file descriptor on disconnection or closeStream.
The FLV class has a reader task, which does not get terminated when the
NetConnection or NetStream is closed in rtmpclient.py. Check whether it works
on rtmp.py.
Original issue reported on code.google.com by kundan10 on 16 Apr 2011 at 1:46
481 response
sometimes on SIP trace I can see 481 response. but everything works fine. What
does it mean ? how to resolve it ?
thanks
Original issue reported on code.google.com by [email protected] on 9 Feb 2011 at 3:50
re-INVITE handling
Currently siprtmp does not support re-INVITE handling (send '501 re-INVITE not
implemented'). Require this feature to modify SDP (for change in remote media
destination or hold/unhold).
Original issue reported on code.google.com by [email protected] on 29 Mar 2011 at 12:24
Add redirectConnection in rtmplite's Client
Reported by Wai Keen Woon <[email protected]>on Sep 15, 2010
Hello, I would like to contribute a patch for RTMP redirection as
described in
http://www.adobe.com/livedocs/flashmediaserver/3.0/hpdocs/help.html?content=0000
0251.html
Index: rtmp.py
===================================================================
--- rtmp.py (revision 4753)
+++ rtmp.py (working copy)
@@ -657,6 +657,15 @@
description=reason, details=None))
self.writeMessage(response.toMessage())
+ def redirectConnection(self, url, reason=''):
+ '''Method to redirect an incoming client to the given url.'''
+ response = Command()
+ response.id, response.name, response.type = 1, '_error', Message.RPC
+ extra = dict(code=302, redirect=url)
+ response.setArg(dict(level='status',
code='NetConnection.Connect.Rejected',
+ description=reason, details=None, ex=extra))
+ self.writeMessage(response.toMessage())
+
def call(self, method, *args):
'''Call a (callback) method on the client.'''
cmd = Command()
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 3:52
timestamps get reset to 0
What steps will reproduce the problem?
1. play an FLV file where two frames have the same absolute timestamp
2. it will send a packet with absolute timestamp=0
3.
What is the expected output? What do you see instead?
The send routine only gets a relative timestamp passed to it. It should
only send an absolute timestamp for the first packet in a stream.
What version of the product are you using? On what operating system?
4.0 Linux
Please provide any additional information below.
The bug is caused by this "or header.time==0" condition at rtmp.py:316
>>>
# now figure out the header data bytes
if header.streamId != message.streamId or header.time == 0:
header.streamId, header.type, header.size, header.time =
message.streamId, message.type, message.size, message.time
control = Header.FULL
<<<
The attached patch seems to fix it for me, by insuring that header.time
will never be 0 after the first packet in the stream.
Original issue reported on code.google.com by [email protected] on 17 Dec 2009 at 8:54
Attachments:
Bad file descriptor error in siprtmp when receiving call
Reported by Nico <[email protected]> on Aug 30, 2010
Hi, I'm trying to use rtmplite for a project and whenever I make a call to the
flash client, I get an error like the one below, I was wondering if you know
how to fix it, although it looks like it is a problem with p2p-sip. Thanks!
rtmp-accept
create answer failed to create an answer
receive RTP exception (<class 'socket.error'>, error(9, 'Bad file descriptor'),
<traceback object at 0xb7bcb4dc>)
Traceback (most recent call last):
File "/root/downloads/p2p-sip/src/std/rfc3550.py", line 689, in receiveRTP
data, remote = yield multitask.recvfrom(sock, self.maxsize)
File "/root/downloads/rtmplite/multitask.py", line 544, in recvfrom
return FDAction(sock, sock.recvfrom, args, kwargs, read=True)
File "/root/downloads/rtmplite/multitask.py", line 409, in __init__
super(FDAction, self).__init__(fd, read, write, exc, timeout)
File "/root/downloads/rtmplite/multitask.py", line 299, in __init__
self.fd = (fd if _is_file_descriptor(fd) else fd.fileno())
File "<string>", line 1, in fileno
File "/opt/python-2.6.6/lib/python2.6/socket.py", line 167, in _dummy
raise error(EBADF, 'Bad file descriptor')
error: [Errno 9] Bad file descriptor
receive RTCP exception (<class 'socket.error'>, error(9, 'Bad file
descriptor'), <traceback object at 0xb7bcb464>)
receive RTP exception (<class 'socket.error'>, error(9, 'Bad file descriptor'),
<traceback object at 0xb7bcb4dc>)
Traceback (most recent call last):
File "/root/downloads/p2p-sip/src/std/rfc3550.py", line 689, in receiveRTP
data, remote = yield multitask.recvfrom(sock, self.maxsize)
File "/root/downloads/rtmplite/multitask.py", line 544, in recvfrom
return FDAction(sock, sock.recvfrom, args, kwargs, read=True)
File "/root/downloads/rtmplite/multitask.py", line 409, in __init__
super(FDAction, self).__init__(fd, read, write, exc, timeout)
File "/root/downloads/rtmplite/multitask.py", line 299, in __init__
self.fd = (fd if _is_file_descriptor(fd) else fd.fileno())
File "<string>", line 1, in fileno
File "/opt/python-2.6.6/lib/python2.6/socket.py", line 167, in _dummy
raise error(EBADF, 'Bad file descriptor')
error: [Errno 9] Bad file descriptor
receive RTCP exception (<class 'socket.error'>, error(9, 'Bad file
descriptor'), <traceback object at 0xb7bcb464>)
sending[307] to ('74.55.36.250', 5060)
SIP/2.0 488 Incompatible SDP
---
Hi Nico,
This might be a bug in the code that both sides are not using the same
SDP session. I see the following error: "SIP/2.0 488 Incompatible SDP"
Can you enable trace log using "-d" option or setting "_debug = True"
in appropriate .py files, so that it prints the full SIP messages.
Then send me the sender and receiver's INVITE and 200 OK message
traces. That will give good deal of information about why SDP m= lines
are incompatible.
Keep in touch and good luck!
---
Thank you very much for your reply, here's the whole trace, from when I start
the server to when the call is terminated with the error. The flow is as
follows:
* Start siprtmp
* Connect with flash
* Originate a call to the extension registered by flash through siprtmp
* When the called is answered in flash, siprtmp throughs an error and send an
SDP saying incompatible destination
This happens only when calling flash. If I make the call from flash, there's no
problem.
received[1139] from ('myserver_ip', 5060)
INVITE sip:[email protected]:60640 SIP/2.0
Via: SIP/2.0/UDP myserver_ip;rport;branch=z9hG4bKrg9D9XB2BXpvg
Max-Forwards: 70
From: "FreeSWITCH" <sip:0000000000@myserver_ip>;tag=r3474y38cjBcj
To: <sip:[email protected]:60640>
Call-ID: cf7b348c-332c-122e-688d-001d09ef4a24
CSeq: 1488363 INVITE
Contact: <sip:mod_sofia@myserver_ip:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15979
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 247
X-FS-Support: update_display
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000@myserver_ip>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1283617883 1283617884 IN IP4 myserver_ip
s=FreeSWITCH
c=IN IP4 myserver_ip
t=0 0
m=audio 30972 RTP/AVP 98 101 13
a=rtpmap:98 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
createServer <UserAgent call-id=cf7b348c-332c-122e-688d-001d09ef4a24>
sending[280] to ('myserver_ip', 5060)
SIP/2.0 100 Trying
Content-Length: 0
Via: SIP/2.0/UDP myserver_ip;rport=5060;branch=z9hG4bKrg9D9XB2BXpvg
From: "FreeSWITCH" <sip:0000000000@myserver_ip>;tag=r3474y38cjBcj
To: <sip:[email protected]:60640>
CSeq: 1488363 INVITE
Call-ID: cf7b348c-332c-122e-688d-001d09ef4a24
receivedRequest method= INVITE ua= <UserAgent
call-id=cf7b348c-332c-122e-688d-001d09ef4a24> for ua without queue
incominghandler connect
Gateway[0] onResult None
Gateway[0] onPublish sip/[email protected] local
Gateway[0] onPlay sip/[email protected] remote
rtmp-accept
create answer failed to create an answer
receive RTP exception (<class 'socket.error'>, error(9, 'Bad file descriptor'),
<traceback object at 0xb7b47414>)
Traceback (most recent call last):
File "/root/downloads/p2p-sip/src/std/rfc3550.py", line 689, in receiveRTP
data, remote = yield multitask.recvfrom(sock, self.maxsize)
File "/root/downloads/rtmplite/multitask.py", line 544, in recvfrom
return FDAction(sock, sock.recvfrom, args, kwargs, read=True)
File "/root/downloads/rtmplite/multitask.py", line 409, in __init__
super(FDAction, self).__init__(fd, read, write, exc, timeout)
File "/root/downloads/rtmplite/multitask.py", line 299, in __init__
self.fd = (fd if _is_file_descriptor(fd) else fd.fileno())
File "<string>", line 1, in fileno
File "/opt/python-2.6.6/lib/python2.6/socket.py", line 167, in _dummy
raise error(EBADF, 'Bad file descriptor')
error: [Errno 9] Bad file descriptor
receive RTCP exception (<class 'socket.error'>, error(9, 'Bad file
descriptor'), <traceback object at 0xb7b4716c>)
receive RTP exception (<class 'socket.error'>, error(9, 'Bad file descriptor'),
<traceback object at 0xb7b473ec>)
Traceback (most recent call last):
File "/root/downloads/p2p-sip/src/std/rfc3550.py", line 689, in receiveRTP
data, remote = yield multitask.recvfrom(sock, self.maxsize)
File "/root/downloads/rtmplite/multitask.py", line 544, in recvfrom
return FDAction(sock, sock.recvfrom, args, kwargs, read=True)
File "/root/downloads/rtmplite/multitask.py", line 409, in __init__
super(FDAction, self).__init__(fd, read, write, exc, timeout)
File "/root/downloads/rtmplite/multitask.py", line 299, in __init__
self.fd = (fd if _is_file_descriptor(fd) else fd.fileno())
File "<string>", line 1, in fileno
File "/opt/python-2.6.6/lib/python2.6/socket.py", line 167, in _dummy
raise error(EBADF, 'Bad file descriptor')
error: [Errno 9] Bad file descriptor
receive RTCP exception (<class 'socket.error'>, error(9, 'Bad file
descriptor'), <traceback object at 0xb7b4743c>)
sending[307] to ('myserver_ip', 5060)
SIP/2.0 488 Incompatible SDP
Content-Length: 0
Via: SIP/2.0/UDP myserver_ip;rport=5060;branch=z9hG4bKrg9D9XB2BXpvg
From: "FreeSWITCH" <sip:0000000000@myserver_ip>;tag=r3474y38cjBcj
To: <sip:[email protected]:60640>;tag=109646747812
CSeq: 1488363 INVITE
Call-ID: cf7b348c-332c-122e-688d-001d09ef4a24
received[325] from ('myserver_ip', 5060)
ACK sip:[email protected]:60640 SIP/2.0
Via: SIP/2.0/UDP myserver_ip;rport;branch=z9hG4bKrg9D9XB2BXpvg
Max-Forwards: 70
From: "FreeSWITCH" <sip:0000000000@myserver_ip>;tag=r3474y38cjBcj
To: <sip:[email protected]:60640>;tag=109646747812
Call-ID: cf7b348c-332c-122e-688d-001d09ef4a24
CSeq: 1488363 ACK
Content-Length: 0
Gateway[0] onClose sip/[email protected] local
Gateway[0] onStop sip/[email protected] remote
Gateway[0] onResult None
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 2:24
Call transfer in siprtmp
Reported by ravindimantha <[email protected]>on Nov 24, 2010.
Hi great project!
Just wanted to know if in the rtmplite project if you have implemented
the Call Transfer function. I ask this question here because the
rtmplite is said to be using the p2p-sip stack from here.
Thanks in advance.
Ravin
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 5:18
playback problem version 7.2
Please describe the issue and also put debug trace. Additionally, please
describe your network configuration (NAT, IP addresses, etc).
Hi,
I've updated my rtmplite versione to the latest 7.2 one, but webcam recorded
clips won't playback correctly - buggy sound and slow video output.
After switching back to 7.0 the playback worked fine.
Original issue reported on code.google.com by [email protected] on 7 Mar 2011 at 12:31
Cannot dial sip address with * in front
Reported by Franck <[email protected]> on Aug 3, 2010
there is this message error when I try to call a sip address with an asterisk
in front :
rtmp-invite *[email protected]
exception in invite (<type 'exceptions.AttributeError'>,
AttributeError("'NoneType' object has no attribute 'dup'",), <traceback object
at 0x1f92b00>)
Gateway[0] onResult None
tried to correct it but my python level is not yet skillfull
Thanks
Franck
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 12:18
Asterisk / SIP with rtmplite problems
I am trying to interconnect SIP and flash using Asterisk as the SIP server, on
my LAN. Here is one of the tests I have done, and its log.
I am using Ekiga, it connects to Asterisk fine and can make calls from and to
other SIP clients.
- First I connect the flash VideoPhone application to rtmplite
- Then I make Ekiga call the VideoPhone
- Then I pick up on the VideoPhone. It instantly connects and disconnects.
Ekiga believes it is still connected, using speex. After 7 seconds, Ekiga hangs
up automatically.
- At this time the VideoPhone starts ringing again (as if it is receiving a
call), but picking up doesn't do anything of course, except reset the
VideoPhone to standby mode, waiting for new calls.
Notice the same symptoms seem to happen when doing flash to flash calls using
asterisk (but not p2psip, there it works fine), flash to phone and phone to
flash (p2psip and asterisk), but asterisk with VideoPhone and Android native
Gingerbread SIP client works fine, and so does Ekiga with the Android SIP
client.
The log of the above described situation (Ekiga calls VideoPhone) is in the
attached file (inlined), as well as the log of a flash to flash call over
asterisk (both are using videophones) with the following scenario:
- First I register the first compouter (1001)
- Next I register the second computer (1004)
- Then 1001 calls 1004
- 1004 picks up and instantly connection is cut
Thanks for any help anyone may provide,
Yannack
Original issue reported on code.google.com by [email protected] on 7 Jul 2011 at 9:33
Attachments:
Audio quality problem with speex8 and Asterisk
Reported by Taras S <[email protected]> on Aug 5, 2010
see
http://groups.google.com/group/myprojectguide/browse_thread/thread/4449bb7c9b48f
bae/cb25671d2e0acd39?pli=1
hi
I used siprtmp and I have a question about it
If I'm calling(using speex8) from flash to asterisk, on music on hold
extensions, sound quality is awful.(in trafshow seen that rtp flow is
unstable. From time to time the flow stops and then resumes again,
this leads to failures in the sound). But if i calling(using speex8)
from flash to freeswitch, on music on hold extensions, sound quality
are normal(steady stream of rtp). I'm confused.
--
There were some timestamp related bug fixes done in version 6.0 of
rtmplite/siprtmp. Can you please give it a try.
If you are already trying version 6.0 or later, could you please
enable RTP trace (set _debug=True in siprtmp.py, rtmplite.py and in
p2p-sip's rfc3550.py) and send the trace as reply. Basically I want to
see the RTP timestamp and actual RTMP time when packets are sent by
asterisk to Flash application.
Thanks.
---
hi
This is tarce log siprtmp ver6.0
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 2:18
about new params --int-ip and --ext-ip
I tried to configure siprtmp with
--int-ip as 127.0.0.1 and --ext-ip as a public ip 12.34.56.78
all SIP messages are going and received well, but no sound.
is it a NAT issue ? if yes, how should I set it ?
Thanks
Franck
Original issue reported on code.google.com by [email protected] on 14 Apr 2011 at 5:37
Performance measurement
Measure the performance of rtmplite for simple (1) video recording, (2)
playback streaming, (3) audio only, (4) siprtmp translation for audio call, (5)
real-time two-party call.
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 7:07
Stream switcher using rtmplite
First reported by Fabio Di Bernardini [email protected] on Tue, Jul 26, 2011
---
I need a simple streaming system where only one client at a time can
stream (due to bandwitdh issues) and a director can choose which
client.
Look at this sketch: http://goo.gl/8vauk
At the end of this system there is a converter and a streaming service
(like ustream, icecast or similar) needed to broadcast the choosed
client.
What do you think? Is it feasible with rtmplite?
---
Hi Fabio,
Yes, this should be feasible with rtmplite.
Assuming that it is your web client, and the simplest option would to
be to send RPC commands from rtmplite to your Flash application to
enable/disable publish of the stream. The Flash client application
connects to the server rtmp://server/myapp/scope123. But the Flash
client will not publish until it receives "start_stream" command from
the server. The server built using rtmp.py and rtmpclient.py creates a
new FlashServer instance to listen to incoming Flash client
connections. The server already keeps track of list of connected
clients, and can decide which client it wants to start streaming from.
Then it sends a "stop_stream" command to any previously active client,
and "start_stream" command to new active client, using the "call"
method (look for how self.client.call is called in siprtmp.py to
indicate an event from server to client). The client application
invokes publish/close based on when it receives "start_stream" or
"stop_stream" commands. If you are using a third party Flash client
application, then you will need to handle this signaling out of band,
e.g., using separate signaling channel.
On the other side of the system, you will need to use the
NetConnection/NetStream objects of rtmpclient.py module to connect to
your existing streaming server, or have the existing streaming server
connect to rtmpclient. The rtmpclient module is capable of connecting
to existing RTMP server to publish or play a stream. You can use the
onPublishData callback (see rtmp.py) to receive the audio data from
the client, and NetStream's send to send it to the other streaming
service.
Original issue reported on code.google.com by kundan10 on 27 Jul 2011 at 4:55
Uplink with ffmpeg: playback with ffplay, vlc and rtmpdump fails
I uplink using ffmpeg...
ffmpeg -i somevideo.mp4 -re -vcodec copy -acodec copy -f flv
rtmp://localhost/live/livestream
In this case the video is H264/MP3 and known to work with other live rtmp
streaming software e.g. Wowza and rtmpd.
If I try to connect to the stream using VLC using...
vlc -vvv rtmp://localhost/live/livestream
I get the following error report...
[0x1b6df00] main input debug: `rtmp://localhost/live/livestream' gives access
`rtmp' demux `' path `localhost/live/livestream'
[0x1b6df00] main input debug: creating demux: access='rtmp' demux=''
path='localhost/live/livestream'
[0x1f3a810] main demux debug: looking for access_demux module: 0 candidates
[0x1f3a810] main demux debug: no access_demux module matched "rtmp"
[0x1f3a810] main demux debug: TIMER module_need() : 0.165 ms - Total 0.165 ms /
1 intvls (Avg 0.165 ms)
[0x1b6df00] main input debug: creating access 'rtmp'
path='localhost/live/livestream'
[0x1f3a9c0] main access debug: looking for access module: 1 candidate
[0x1f3a9c0] access_avio access debug: Opening 'rtmp://localhost/live/livestream'
Warning: call to rand()
Warning: call to rand()
Warning: call to rand()
Warning: call to rand()
Closing connection: NetStream.Play.StreamNotFound
[0x1f3a9c0] access_avio access error: Failed to open url using libavformat
[0x1f3a9c0] main access debug: no access module matching "rtmp" could be loaded
[0x1f3a9c0] main access debug: TIMER module_need() : 81.128 ms - Total 81.128
ms / 1 intvls (Avg 81.128 ms)
[0x1b6df00] main input error: open of `rtmp://localhost/live/livestream'
failed: (null)
If I open with ffplay...
ffplay rtmp://localhost/live/livestream
I get the following error report...
Closing connection: NetStream.Play.StreamNotFound
rtmp://localhost/live/livestream: Operation not permitted
If I connect using rtmpdump piped into ffplay...
rtmpdump -r rtmp://localhost/live/livestream -q -o - -v | ffplay -
I get only sound, no picture, and the following error report...
[h264 @ 0x22ccc80]no frame!
[flv @ 0x22a86b0]invalid stream
[h264 @ 0x22ccc80]no frame!
Last message repeated 4 times
[h264 @ 0x22ccc80]slice type too large (0) at 0 0
[h264 @ 0x22ccc80]decode_slice_header error
[h264 @ 0x22ccc80]no frame!
(repeated)
Original issue reported on code.google.com by [email protected] on 22 Feb 2011 at 3:19
While label phone for siprtmp
Reported by <[email protected]>on Oct 7, 2010
Hello, and Congratulations for your nice little software SIP
videophone.
I would like to do some experiments with it and I need your help
for that as soon as possible.
1- I would like a plain version of the flash-to-Sip videophone page
without: (the text input form, without the buttons, without the
tool-tips, without the confirmation yellow bars in the bottom, just
a white background, no border, in other words, no visual elements
at all, but only a white plain square SWF object that I can fix the
size and the loggin infos in the html source myself). I ask you
because I don't want to break anything in the asc code if I edit it
myself for my experiments. send it in my email Please, I'm willing
to give you a donation for your help, and for future experiments
help also :-)
Best Regards
Marco
---
Hi Marco,
Not sure if we replied to you earlier or not. So am replying again.
Were you able to do this yourself?
You can edit View.mxml to remove various UI elements.
Then you can add code in VideoPhone.mxml::installJavaScriptAPI() to
add more API to control from JavaScript.
If you haven't done it already and if you find it difficult, I will
try to send you the new SWF. Let me know.
---
Hi Kundan,
I cannot do it, I was still waiting for your help, my experiment
need is that I will make it call a SIP number, that's it. so I
don't want
no graphics at all, and also I want it to be able to call just a SIP
number, without password or name remembering.
Can you do it for me please?
Best Regards
Marco
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 4:10
Changing Message to Message(object) breaks
Reported by Valery <[email protected]> on Nov 30, 2009.
> Hi,
>
> Most probably there is a bug in your code.
> If I change Message to class Message(object), playing stop working. Second
> connection hangs and response is not sent back to flash.
>
> The issue somewhere in property, fget, fset. Without 'object', message.type
> does not set message.header.type and everything works.
>
> So this is strange for me.
>
> It will be great if you explain that and provide a fix.
>
> Thanks,
> Valery.
>
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 7:13
In about 2-5 minutes, video freezes
Reported by Valery <[email protected]> on Dec 9, 2009.
I got the latest source code and I found another issue, that seems is more
critical.
In about 2-5 minutes of playing, the picture freezes. I dig a little and found
out that the recv on publisher's socket stoped getting data.
To reproduce it faster I do minor changes to inclease bitrate.
- I changed width and height for video to 320x240
- added the code to 'private function publish'
ns.publish(name, record ? 'record' : null);
var camera:Camera = Camera.getCamera();
camera.setMode(wVideo, hVideo, 30);
camera.setQuality(0, 100);
local.attachCamera(camera);
ns.attachCamera(camera);
In about 30 seconds, it stops working.
Thanks,
Valery.
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 7:23
Cannot play using FLVPlayback component
Reported by Alexey Eremin <[email protected]> on Feb 17, 2010.
Hello.
Thank you for this project. I try to use it, but i have some question.
I try to record some video by call publish("some", "record"), server
create a file some.flv.
In next step i want to try play it, i write little player, which use
NetStream and Video components - it's works.
But i want to use FLVPlayback to watch recorded video, but it
components can't play video from server.
I try set a source property of component and try use play method, but
it not work.
In server debug mode i see, that server resive connection request and
answer on it, but next component command(s) not sent.
How to use FLVPlayback component?
Thank you. Aleksey.
P.S. Sorry for my english.
Original issue reported on code.google.com by [email protected] on 3 Feb 2011 at 7:37
Attachments:
Audio dropping off with VideoIO.swf and rtmplite
Reported by Diarmuid Wrenne <[email protected]> on Nov 2, 2010.
Hi
Sorry for emailing you directly but I am not sure if the support is
working on Google Groups.
The audio seems to be dropping off when I record a file. The file
attached is an example.
Could the be something to do with the built in mic or some setting I
need to change
Great work on the apps and your write ups.
Cheers
Diarmuid
---
Hi Diarmuid,
Do you have some other audio tool which you can record and playback to
verify that your mic is working fine?
I suspect it is something else. Few questions if you can answer:
1. Is rtmplite running on same host as your browser in your test?
2. If (1) is no, then how is the bandwidth between your browser's host
and the host running rtmplite?
3. Do you have information about CPU usage (either window's task
manager or Mac's activity monitor can give this information) on the
machine that runs rtmplite? and that runs your browser? Is it showing
high (close to 100%) CPU usage?
4. Did you change any default parameter in VideoIO?
5. Which version of rtmplite are you using? or are you using the
latest SVN code?
6. What is your flash player version? Right click in VideoIO and then
left click on 'Adobe Flash Player...' to know the exact version
number.
---
Also I am attaching another VideoIO.swf if you can replace yours with this one.
The difference is that this one disables silence-suppression in
Microphone object of Flash Player, in case this was an issue.
Also in your experiment, if you can set the mic's volume to be max,
that will help.
---
Hi Kundan,
Thanks for the great reply.
Activity Monitor shows rtmplite running at 6 % when recording 1 stream.
Audio works fine with Photobooth.
I set the max volume in the flash recorder and that had the biggest
effect on the audio recorder. I think the echo check box may have had
some affect as well.
I also tested the Flash recorder (not rtmplite) in Facebook that
works the same way with the volume control needing to be very high for
any audio to be recorded consistently. It could be something to do
with the 24" i-mac's build in mic. I also played around with the mac's
built in audio prefs and the default settings work well. Ambient noise
reduction makes a big difference but seems to have little impact on
the recorder
The recorder volume in flash seems to be quite crude as it need to be
very high before it picks up the sound. I wonder is this scriptable on
javascript.
Do you know of a RTMFP server to help with making peer to peer calls.
I see you use the Adobe stratus server , but maybe there is an open
source server out there??
I tried the recoder with two inpute streams and it worked fine. Any
idea how many could be recorded at once or is it based on server
capacity ?
Thanks for your help. I was telling a friend today how brilliant your
code and write-ups are. They are very impressive.
Cheers
---
Hey Diarmuid,
The recorder volume is also scriptable from Javascript using the
"gain" property as described in
http://myprojectguide.org/p/flash-videoio/10.html.
I don't know of any open source rtmfp server. There was some effort
that started out, but I haven't followed it recently.
One of the pending task I need to do is capacity/performance
measurement of rtmplite.
Hopefully I should be able to do that soon.
Thanks!
Original issue reported on code.google.com by [email protected] on 5 Feb 2011 at 4:07
Attachments:
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