Why won't VLC send an rtsp PLAY request to my server? - tcp

I'm implementing an RTSP server in NodeJs, using the RFC's (rtsp, rtp, sdp) and this tutorial.
I'm using VLC to test my implementation, and it works fine for the example (link at the bottom of the tutorial) but stops halfway for my server.
I'm suspecting some RFC compliance issue, but I can't find it, and VLC isn't really providing any useful information as to what it's doing.
Running wireshark and the c++ server implementation, and pointing VLC to it shows all the steps:
OPTIONS rtsp://192.168.10.151:8554/mjpeg/1 RTSP/1.0
CSeq: 2
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
RTSP/1.0 200 OK
CSeq: 2
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
DESCRIBE rtsp://192.168.10.151:8554/mjpeg/1 RTSP/1.0
CSeq: 3
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
Accept: application/sdp
RTSP/1.0 200 OK
CSeq: 3
This should be date
Content-Base: rtsp://192.168.10.151:8554/mjpeg/1/
Content-Type: application/sdp
Content-Length: 90
v=0
o=- 6334 1 IN IP4 192.168.10.151
s=
t=0 0
m=video 0 RTP/AVP 26
c=IN IP4 0.0.0.0
SETUP rtsp://192.168.10.151:8554/mjpeg/1/ RTSP/1.0
CSeq: 4
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
RTSP/1.0 200 OK
CSeq: 4
This should be date
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Session: -2144778205
PLAY rtsp://192.168.10.151:8554/mjpeg/1/ RTSP/1.0
CSeq: 5
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
Session: -2144778205
Range: npt=0.000-
RTSP/1.0 200 OK
CSeq: 5
This should be date
Range: npt=0.000-
Session: -2144778205
RTP-Info: url=rtsp://127.0.0.1:8554/mjpeg/1/track1
And the VLC messages:
...
live555 debug: RTP subsession 'video/JPEG'
core debug: selecting program id=0
live555 debug: setup start: 0.000000 stop:0.000000
live555 debug: play start: 0.000000 stop:0.000000
core debug: using access_demux module "live555"
core debug: looking for decoder module matching "any": 43 candidates
...
When I run my own server, it never sends the play request:
OPTIONS rtsp://rasmus.axit.local:8554/mjpeg/1 RTSP/1.0
CSeq: 2
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
RTSP/1.0 200 OK
CSeq: 2
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
DESCRIBE rtsp://rasmus.axit.local:8554/mjpeg/1 RTSP/1.0
CSeq: 3
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
Accept: application/sdp
RTSP/1.0 200 OK
CSeq: 3
Date: Fri, 09 Sep 2016 09:36:29 GMT
Content-Base: rtsp://rasmus.axit.local:8554/mjpeg/1
Content-Type: application/sdp
Content-Length: 91
v=0
o=- -12345678 1 IN IP4 192.168.10.71
s=
t=0 0
m=video 0 RTP/AVP 26
c=IN IP4 0.0.0.0
SETUP rtsp://rasmus.axit.local:8554/mjpeg/1/ RTSP/1.0
CSeq: 4
User-Agent: LibVLC/2.2.1 (LIVE555 Streaming Media v2014.07.25)
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
RTSP/1.0 200 OK
CSeq: 4
Date: Fri, 09 Sep 2016 09:36:29 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
Session: -12345678
And VLC:
...
live555 debug: RTP subsession 'video/JPEG'
It doesn't continue from there.
I can't figure out what it's missing. Earlier it didn't send a SETUP either, and that turned out to be a missing empty line on the DESCRIBE response. Consequently I've tried adding various amount of newlines, ids, different dates and what not in different places, but no dice.
Please let me know if you need more information.

Your response to the DESCRIBE commands seems to be invalid - you are replying with 96 characters while the Content-Length header states 91. Not sure if this affects the outcome, but I assume VLC could fail because of this (it could be that it cannot parse the connection data line). Also there seems to be an unneeded extra newline at the end of the SDP data at the end of the last line.

Related

Asterisk returns 501 - Not Implemented to INFO packets

I am trying to send DTMFs through SIPp. Since the play_pcap_audio action was not 100% reliable, I wanted to construct SIP INFO messages to make my tests more robust, however when I send INFO packets I get 501 - Not implemented response from Asterisk.
If I set my softphone to use SIP INFO for sending DTMFs, that works fine, so I am assuming it has to do with the messages I am sending. However comparing the actual messages did not reveal any difference.
The INVITE I send:
INVITE sip:*203#192.168.200.208:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.2:5060;branch=z9hG4bK-234-1-7;rport
From: sipp <sip:2018005#192.168.200.208>;tag=1
To: <sip:*203#192.168.200.208:5060>
Call-ID: 1-234#172.17.0.2
CSeq: 4 INVITE
Contact: sip:2018005#172.17.0.2:5060
Authorization: //auth header omitted
Max-Forwards: 70
Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO
Content-Type: application/sdp
Content-Length: 195
v=0
o=user1 53655765 2353687637 IN IP4 172.17.0.2
s=-
c=IN IP4 172.17.0.2
t=0 0
m=audio 8192 RTP/AVP 0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
and the INFO message:
INFO sip:192.168.200.208:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.0.2:5060;branch=z9hG4bK-234-1-16
Max-Forwards: 70
Contact: <sip:2018005#172.17.0.2:5060;transport=UDP>
To: <sip:*203#192.168.200.208:5060>
From: "sipp" <sip:2018005#192.168.200.208>;tag=1
Call-ID: 1-234#172.17.0.2
CSeq: 5 INFO
User-Agent: SIPp docker
Authorization: // auth header omitted
Content-Length: 22
Signal=1
Duration=160
I have made sure it's not to do with the dtmfmode configuration in Asterisk.
One thing I noticed is when Asterisk responds to INVITE, it contains the following header:
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
I would expect INFO to be here - but again, it's the same when using the softphone and it all works.
What other areas could affect the processing of SIP INFOs?
Any help would be much appreciated on further debugging.
Turned out to be a problem with SIP dialogs.
The tags (+ a call-id) identify a dialog. After an INVITE is sent the UAS responds with an OK, sticking a remote tag to the To: field. Every subsequent request has to use the same tags (local and remote + call-id) to refer to the same dialog. This can be achieved by sticking [last_To:] into the header when using a SIPp scenario, so that we get the correct remote tag:
<send>
<![CDATA[
ACK sip:[field3]#[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: <sip:[field0]#[field1]>;tag=[call_number]
[last_To:]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: sip:[field0]#[local_ip]:[local_port]
Max-Forwards: 10
Content-Length: 0
]]>
</send>
In the above case an ACK is sent back from the UAC and the dialog is established. Now when we send the INFO, we have to refer to the same dialog (by setting the correct tags) and it all works.
Interestingly, when not setting these values properly, pjsip gives 501 Not Implemented, whereas chan_sip responds with 481 Call/Transaction Does Not Exist which is much more accurate.

Asterisk returns 481 on hangup

Not sure if it's the place where I should ask this question.
I'm developing a simple voip application. I can call to other users, but can't hang up. When client sends BYE request, server answers with 481 - call leg transaction does not exist.
Here are client logs:
INVITE sip:2#172.20.4.7:51110;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.20.4.7:5060;branch=z9hG4bK06952c7a;rport
Max-Forwards: 70
From: "First" <sip:1#172.20.4.7>;tag=as746cc61d
To: <sip:2#172.20.4.7:51110;transport=UDP>
Contact: <sip:1#172.20.4.7:5060>
Call-ID: 425bb181009f366c499b10f362d29ac6#172.20.4.7:5060
CSeq: 102 INVITE
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Date: Tue, 07 Mar 2017 11:52:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "First" <sip:1#172.20.4.7>
Content-Type: application/sdp
Content-Length: 310
v=0
o=root 2015785808 2015785808 IN IP4 172.20.4.7
s=Asterisk PBX 11.16.0
c=IN IP4 172.20.4.7
t=0 0
m=audio 13952 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bKnCqg
Contact: <sip:2#172.20.4.7:51110;transport=UDP>
To: <sip:2#172.20.4.7;transport=UDP>;tag=YU2R
From: <sip:2#172.20.4.7;transport=UDP>;tag=as746cc61d
Call-ID: 425bb181009f366c499b10f362d29ac6#172.20.4.7:5060
CSeq: 102 INVITE
Allow-Events: presence, kpml, talk
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bK06952c7a;rport
Contact: <sip:2#172.20.4.7:51110;transport=UDP>
To: <sip:2#172.20.4.7;transport=UDP>;tag=YU2R
From: "First" <sip:1#172.20.4.7;transport=>;tag=as746cc61d
Call-ID: 425bb181009f366c499b10f362d29ac6#172.20.4.7:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Accept: application/sdp, application/sdp
Accept-Language: en
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
Allow-Events: presence, kpml, talk
Content-Length: 247
v=0
o=Z 13 13 IN IP4 95.86.129.80
s=Test
c=IN IP4 95.86.129.80
t=0 0
m=audio 50000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode = 30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
ACK sip:2#172.20.4.7:51110;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.20.4.7:5060;branch=z9hG4bK54ba94a0;rport
Max-Forwards: 70
From: "First" <sip:1#172.20.4.7>;tag=as746cc61d
To: <sip:2#172.20.4.7:51110;transport=UDP>;tag=YU2R
Contact: <sip:1#172.20.4.7:5060>
Call-ID: 425bb181009f366c499b10f362d29ac6#172.20.4.7:5060
CSeq: 102 ACK
User-Agent: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Content-Length: 0
BYE sip:1#172.20.4.7;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bKAZsQ
Max-Forwards: 70
From: <sip:2#172.20.4.7;transport=UDP>;tag=as746cc61d
To: <sip:1#172.20.4.7;transport=UDP>;tag=YU2R
Contact: <sip:2#172.20.4.7:51110;transport=UDP>
CSeq: 2 BYE
User-Agent: TestSoftphone
Call-ID: 425bb181009f366c499b10f362d29ac6#172.20.4.7:5060
Content-Length: 0
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 172.20.4.7:51110;branch=z9hG4bKAZsQ;received=172.20.1.40;rport=51110
From: <sip:2#172.20.4.7;transport=UDP>;tag=as746cc61d
To: <sip:1#172.20.4.7;transport=UDP>;tag=YU2R
Call-ID: 425bb181009f366c499b10f362d29ac6#172.20.4.7:5060
CSeq: 2 BYE
Server: FPBX-AsteriskNOW-12.0.76.4(11.16.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
I read that 481 can occur if To tag or From tag or Call-ID is outside of dialog, but looks like tags and ids are ok. I don't receive a tag for To, so I generate Ringing packet where To tag is set, that may be the problem.
EDIT. I added "pedantic=no" to sip_custom.conf, and it works now. Though I don't know what's wrong with this dialog.
I'm pretty sure that Contact header is not necessary in BYE. Fastest way to understand what is wrong is check full dialog with some other SIP client. As you sad this is probably related with To or From headers
Please compare the content of Via headers in initial INVITE and BYE messages - values are different: Via: SIP/2.0/UDP 172.20.4.7:5060 and Via: SIP/2.0/UDP 172.20.4.7:51110 respectively. Asterisk created dialog and marked it internally (inside its mind) according to combination of address and port in Via header. It expects actual dialog to be canceled from the same address and port, hence all the requests came from different source are considered as "out-of-dialog" and rejected.
Actual problem is widespread and caused by incorrect behavior of the routers/firewalls installed on the edge of client networks. Try to disable ALG feature on mentioned device.

Asterisk Instant Messaging Errors

I was trying to setup instant messaging in asterisk server. For the client i'm using Blink Softphone. I did add to my sip.conf
[general]
accept_outofcall_message=yes
outofcall_message_context=dialplan_name
auth_message_requests=yes
and to my extensions.conf
[dialplan_name]
exten => _XXX,1,MessageSend(sip:${EXTEN},"${CALLERID(name)}"${MESSAGE(from)})
So this is a simple extension for testing. But when i try and send the message from user1 to user2 i get in the asterisk log:
[Jan 30 21:17:47] WARNING[6420][C-00000005]: chan_sip.c:10515 process_sdp: Insufficient information in SDP (c=)...
What can be wrong here? I'am sure that the clients are on the nat, so i do have both nat=force_rport,comedia for all my users in sip.conf
My asterisk version is 13 (latest).
[UPDATED]
I turned on the sip debug log, and tried to send the message (first i get some weird retransmission):
Retransmitting #9 (no NAT) to 14.228.14.150:5070: <- weird ip here
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 14.228.14.150:5070;branch=z9hA4cA-abcdef613aca8b1x3124abb0z3e1bc8;received=14.228.14.150;rport=5070
From: 2014<sip:2014#(server_ip_here)>;tag=a312facc
To: 0009735466221178<sip:0009735466221178#(server_ip_here)>;tag=bb62441233
Call-ID: abcdef613aca8b1x3124abb0z3e1bc8
CSeq: 1 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2222abcf"
Content-Length: 0
Then
<--- SIP read from TLS:(my_ip_here):49312 --->
INVITE sip:123#(server_ip_here) SIP/2.0
Via: SIP/2.0/TLS (my_ip_here):49312;rport;branch=b4ae88b115be15a5n9244;alias
Max-Forwards: 70
From: "user1" <sip:user1#server_ip_here>;tag=39e388fd5f616b7
To: <sip:123#(server_ip_here)>
Contact: <sip:12313560#(server_ip_here):49311;transport=tls>
Call-ID: ac48128192za12a2f432e24c18aabc7q
CSeq: 28982 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.9.1.2 (Windows)
Content-Type: application/sdp
Content-Length: 308
v=0
o=- 3211422221 3211422221 IN IP4 (my_ip)
s=Blink 0.9.1.2 (Windows)
t=0 0
m=message 2855 TCP/TLS/MSRP *
c=IN IP4 (my_ip)
a=path:msrps://(my_ip):2855/37a1cc82ab315e21b222;tcp
a=accept-types:message/cpim text/* application/im-iscomposing+xml
a=accept-wrapped-types:*
a=setup:active
<------------->
--- (13 headers 10 lines) ---
Sending to (my_ip):49312 (NAT)
Sending to (my_ip):49312 (NAT)
Using INVITE request as basis request - ad72cd1681aff769721af12c21aaea7c
Found peer 'user1' for 'user1' from (my_ip):49312
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
[Jan 31 21:02:17] WARNING[27265][C-00000118]: chan_sip.c:10515 process_sdp: Insufficient information in SDP (c=)...
arheops was right! Using INVITE request as basis request <- here it is. Instead of MESSAGE for some reason INVITE request is used.
Looks like your softphone use INVITE instead of MESSAGE for messaging.
You can get more info by enable sip debug in asterisk console
asterisk -r
sip set debug on

IceCast 2.3.2-kh29 server streaming 404 Error

I am loading a MP3 stream from IceCast 2.3.2-kh29 server in the Android app with MediaPlayer class.
Playing works well, but sometimes stops happen. If see the server responses in IcyStreamMeta class for ID3 tags, there is 404 error for this case.
Also it happens in Windows 7: Firefox and other browsers.
Here are normal headers (some data ***ed):
http://***:14534/***.mp3
GET /***.mp3 HTTP/1.1
Host: ***:14534
User-Agent: Mozilla/5.0 (Windows NT 6.1; WOW64; rv:22.0) Gecko/20100101 Firefox/22.0
Accept: text/html,application/xhtml+xml,application/xml;q=0.9,*/*;q=0.8
Accept-Language: ru-RU,ru;q=0.8,en-US;q=0.5,en;q=0.3
Accept-Encoding: gzip, deflate
Connection: keep-alive
HTTP/1.1 200 OK
Server: nginx/1.4.1
Date: Tue, 23 Jul 2013 21:22:00 GMT
Content-Type: audio/mpeg
Transfer-Encoding: chunked
Connection: keep-alive
icy-br: 192
ice-audio-info: bitrate=192;samplerate=44100;channels=2
icy-description: MP3 192 Kbps
icy-genre: ***
icy-name: ***
icy-pub: 1
icy-url: ***
Cache-Control: no-cache
Expires: Mon, 26 Jul 1997 05:00:00 GMT
Pragma: no-cache
So, the stream sometimes plays only about a minute or less, sometimes seconds and stops. What's the possible reason of 404 error? In other devices there were tests with stable work. Internet speed is well. Can router cause such things? Also, maybe some special HTTP request headers are needed for IceCast (and if they're not present, it gives 404)? Or it's an internal server error for some cases?
So, from WireShark:
2973 53.630385000 SERVER'S IP 192.168.100.6 TCP 1466 14534 > 59847 [ACK] Seq=1284017 Ack=1 Win=63 Len=1412
2976 53.636352000 SERVER'S IP 192.168.100.6 TCP 1157 14534 > 59847 [PSH, ACK] Seq=1285429 Ack=1 Win=63 Len=1103
2978 53.671606000 SERVER'S IP 192.168.100.6 TCP 60 14534 > 59847 [PSH, ACK] Seq=1286532 Ack=1 Win=63 Len=5
2980 53.678606000 SERVER'S IP 192.168.100.6 TCP 60 14534 > 59847 [FIN, ACK] Seq=1286537 Ack=2 Win=63 Len=0
The issue is your chunked encoding. You're proxying your stream through Nginx, and Nginx is "fixing" the output to be compatible with HTTP/1.0. Don't do that.
You can try turning off chunked encoding in your Nginx config:
chunked_transfer_encoding off

Unable to stream RTSP using Live555 proxy server

I am using Live555 streaming media for an application which records and re-streams RTSP streams coming from IP camera. For that, I am using openRTSP for recording and live555 proxy server for re-streaming the camera stream. For a few of the cameras we are facing a strange issue where in the camera recording happens successfully, however the live555 proxy server is unable to generate a new stream for the same camera stream (there is no indication of failure in the verbose output dump, however the rtsp url generated by proxy server cannot be decoded by an rtsp client). Since I do not have any idea about the live555 proxy server details, I have been unable to get into this problem. I tried streaming the same camera stream using VLC and that works fine. What could be possibly wrong with this. I am hereby attaching the verbose output for reference.
E:\...\live\proxyServer>live555ProxyServer.exe -V rtsp://10.17.10.67/ch0_unicast_firststream
LIVE555 Proxy Server
(LIVE555 Streaming Media library version 2012.05.17)
Opening connection to 10.17.10.67, port 554...
RTSP stream, proxying the stream "rtsp://10.17.10.67/ch0_unicast_firststream"
Play this stream using the URL "rtsp://10.17.1.150/proxyStream"
(We use port 8000 for optional RTSP-over-HTTP tunneling.)
...remote connection opened
Sending request: DESCRIBE rtsp://10.17.10.67/ch0_unicast_firststream RTSP/1.0
CSeq: 2
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Accept: application/sdp
Received 716 new bytes of response data.
Received a complete DESCRIBE response:
RTSP/1.0 200 OK
CSeq: 2
Date: Wed, Jul 04 2012 10:55:19 GMT
Content-Base: rtsp://10.17.10.67/ch0_unicast_firststream/
Content-Type: application/sdp
Content-Length: 540
v=0
o=- 1341385393116860 1 IN IP4 10.17.10.67
s=Session of first stream
i=First Codec Stream
t=0 0
a=tool:LIVE555 Streaming Media v2007.08.03
a=type:broadcast
a=control:*
a=range:npt=0-
a=x-qt-text-nam:Session of first stream
a=x-qt-text-inf:First Codec Stream
m=video 0 RTP/AVP 96
c=IN IP4 0.0.0.0
a=rtpmap:96 MP4V-ES/90000
a=fmtp:96 profile-level-id=5;config=000001B005000001B509000001000000012000847A98
28A02240A31F
a=control:track1
m=metadata 0 RTP/AVP 97
c=IN IP4 0.0.0.0
a=rtpmap:97 METADATA/64000
a=control:track2
ProxyServerMediaSession["rtsp://10.17.10.67/ch0_unicast_firststream/"] added new
"ProxyServerMediaSubsession" for RTP/video/MP4V-ES track
ProxyServerMediaSession["rtsp://10.17.10.67/ch0_unicast_firststream/"] added new
"ProxyServerMediaSubsession" for RTP/metadata/METADATA track
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 3
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 122 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 3
Date: Wed, Jul 04 2012 10:55:56 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Opening connection to 10.17.10.67, port 554...
...remote connection opened
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 4
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 122 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 4
Date: Wed, Jul 04 2012 10:56:48 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Opening connection to 10.17.10.67, port 554...
...remote connection opened
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 5
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 122 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 5
Date: Wed, Jul 04 2012 10:57:43 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 6
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 122 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 6
Date: Wed, Jul 04 2012 10:58:23 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 7
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 122 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 7
Date: Wed, Jul 04 2012 10:59:04 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 8
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
ProxyRTSPClient["rtsp://10.17.10.67/ch0_unicast_firststream/"]: lost connection
to server ('errno': 10057). Resetting...
Opening connection to 10.17.10.67, port 554...
...remote connection opened
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 9
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 122 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 9
Date: Wed, Jul 04 2012 11:00:29 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Opening connection to 10.17.10.67, port 554...
...remote connection opened
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 10
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 123 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 10
Date: Wed, Jul 04 2012 11:01:22 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 11
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 123 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 11
Date: Wed, Jul 04 2012 11:02:05 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 12
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 123 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 12
Date: Wed, Jul 04 2012 11:02:39 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 13
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 123 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 13
Date: Wed, Jul 04 2012 11:03:10 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Sending request: OPTIONS rtsp://10.17.10.67/ch0_unicast_firststream/ RTSP/1.0
CSeq: 14
User-Agent: ProxyRTSPClient (LIVE555 Streaming Media v2012.05.17)
Received 123 new bytes of response data.
Received a complete OPTIONS response:
RTSP/1.0 200 OK
CSeq: 14
Date: Wed, Jul 04 2012 11:03:46 GMT
Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Awaiting your response.
Regards,
Saurabh Gandhi
It could be mostly because of udp ports blocked because of firewalls. Try -t flag to force the transmission through tcp.

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