Re: [SR-Users] Wrong location entries when using usrloc with Mongo
Hi,
The log entries are all of the form:
DEBUG: db_mongodb [mongodb_dbase.c:948]: db_mongodb_delete(): delete filter
document: { "expires" : { "$date" : 1422728781000 }, "expires" : { "$date" : 0
} }
Mickael
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel-Constantin Mierla
Sent: Friday, January 30, 2015 4:28 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Wrong location entries when using usrloc with Mongo
I checked quickly the code and mongo c api, it looks ok. Can you see a debug
message with content like:
... delete filter document: ...
when running with debug=3? If yes, can you send it over to check if is correct?
Cheers,
Daniel
On Fri, Jan 30, 2015 at 10:45 AM, Daniel-Constantin Mierla
wrote:
Hello,
it seems that the fileds inside the object are deleted, not the entire object.
The match was done on username and ruid for deletion, both of them are missing.
I will look at the mongo api to see if something was set wrong there for the
delete command.
Cheers,
Daniel
On Fri, Jan 30, 2015 at 9:54 AM, Mickael Marrache
wrote:
I forgot to precise that I allow only one contact per AOR.
modparam("registrar", "max_contacts", 1)
save("location", "0x04")
From: Mickael Marrache [mailto:mickaelmarra...@gmail.com]
Sent: Friday, January 30, 2015 10:51 AM
To: sr-users@lists.sip-router.org
Subject: Wrong location entries when using usrloc with Mongo
Hi,
I start with no location nor in Mongo, nor in memory. My UA registers
successfully and I can see the location entry in Mongo. Then, I close my UA
which unregisters (by setting Expires header to 0). Then, I open the app again
and a new registration is made.
The entry after first registration. Looks okay.
{ "_id" : ObjectId("54cb38f684e58133783f2b42"), "username" : "A", "contact" :
"sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" :
ISODate("2015-01-30T08:55:34Z"), "q" : -1, "callid" :
"297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8809, "flags" : 0,
"cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" :
"sip:XXX:54217;transport=tcp", "path" :
"",
"socket" : "udp:X:5060", "methods" : 4751, "last_modified" :
ISODate("2015-01-30T07:55:34Z"), "ruid" : "uloc-54cb38df-3378-2", "instance" :
null, "reg_id" : 0 }
The same entry after un register (Expires 0). Note that the username field is
missing. In any case, I expected the entry to be deleted.
{ "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" :
ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags"
: 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip:
X:54217;transport=tcp", "path" : "", "socket" : "udp: X:5060",
"methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" :
"297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0,
"contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" }
The entries after second registration. The new entry looks okay. But, the old
entry is still here.
{ "_id" : ObjectId("54cb38f684e58133783f2b42"), "expires" :
ISODate("2015-01-30T08:56:51Z"), "q" : -1, "cseq" : 8811, "flags" : 0, "cflags"
: 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" : "sip:
X:54217;transport=tcp", "path" : "", "socket" : "udp: X:5060",
"methods" : 4751, "last_modified" : ISODate("2015-01-30T07:56:51Z"), "callid" :
"297EC55073A07BF78F0C05822A6CCDFB47E48705", "instance" : null, "reg_id" : 0,
"contact" : "sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp" }
{ "_id" : ObjectId("54cb3a7884e581337a25b895"), "username" : "A", "contact" :
"sip:A@192.168.1.3:54217;rinstance=DFAEBBC7;transport=tcp", "expires" :
ISODate("2015-01-30T09:02:00Z"), "q" : -1, "callid" :
"297EC55073A07BF78F0C05822A6CCDFB47E48705", "cseq" : 8813, "flags" : 0,
"cflags" : 0, "user_agent" : "Acrobits Softphone Business/3.1", "received" :
"sip: X:54217;transport=tcp", "path" : "", "socket" :
"udp: X:5060", "methods" : 4751, "last_modified" :
ISODate("2015-01-30T08:02:00Z"), "ruid" : "uloc-54cb38df-337a-1", "instance" :
null, "reg_id" : 0 }
The issue is the entry is not deleted after un registering.
I precise that registrations are dispatched over multiple registrars with all
accesses the same Mongo cluster.
Thanks,
Mickael
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http://twitter.com/#!/miconda - http://www.linkedin.
[SR-Users] Issue with dlg_setflag vs. setflag
I'm currently using Kamailio built from master@e59db79 and have been following
the recent threads about the dialog module as I'm moving toward adding dialog
support for presence and CDRs in my setup.
However, I seem to have come across an issue where dlg_setflag() does nothing,
while setflag() works properly with the following (trimmed) script. So for
now, I issue setflag(FLT_DIALOG) for INVITE rather than
dlg_setflag(FLT_DIALOG).
Is there something I'm missing about dlg_setflag()?
#!define FLT_DIALOG 4
request_route {
route(REQINIT);
route(NATDETECT);
if(is_method("CANCEL")) {
if(t_check_trans()) {
route(RELAY);
}
exit;
}
route(WITHINDLG);
if(t_precheck_trans()) {
t_check_trans();
exit;
}
t_check_trans();
route(AUTH);
remove_hf("Route");
if(is_method("INVITE|SUBSCRIBE"))
record_route();
if(is_method("INVITE")) {
setflag(FLT_ACC);
# Enable dialog support (dlg_setflag not working)
#dlg_setflag(FLT_DIALOG);
setflag(FLT_DIALOG);
}
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
if($rU==$null) {
# request with no Username in RURI
sl_send_reply("484","Address Incomplete");
exit;
}
route(PSTN);
route(LOCATION);
route(RELAY);
}
route[WITHINDLG] {
if(!has_totag()) return;
if(loose_route()) {
route(DLGURI);
if(is_method("BYE")) {
setflag(FLT_ACC);
setflag(FLT_ACCFAILED);
# Testing dialog flag
xlog("L_INFO", "Completed $dlg(from_uri) to
$dlg(to_uri) - $DLG_lifetime duration\n");
} else if(is_method("ACK")) {
route(NATMANAGE);
} else if(is_method("NOTIFY")) {
record_route();
}
route(RELAY);
exit;
}
if(is_method("SUBSCRIBE") && uri==myself) {
route(PRESENCE);
exit;
}
if(is_method("ACK")) {
if(t_check_trans()) {
route(RELAY);
exit;
} else {
exit;
}
}
sl_send_reply("404","Not Found");
exit;
}
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Re: [SR-Users] What could cause an intermittent fault (28) with http_query()?
What happens if you run the daemon in foreground / high verbosity ? Paste error logs back here. My guess would be there is some kind of blocking happening, ie DNS lookup possibly try using an alternate resolver ? Sent from my iPhone > On Jan 30, 2015, at 10:19 AM, Tim Chubb wrote: > > Hi All > > I seem to be experiencing an intermittent fault with the utils http_query() > method. We are implementing a routing and white list component that is > accessed via a REST api, however i have observed several occasions where this > is logged: > Jan 30 16:04:17 vs-kam-prod02 /usr/local/sbin/kamailio[13184]: ERROR: utils > [functions.c:149]: http_query(): failed to perform curl (28) > > The indicated error number (28) seems to suggest a timeout is occurring with > curl, however examining a capture of network traffic when this happens shows > that a http request is not sent from the server to the destination at all, > usually attempting a second call results in everything working correctly. > > As stated its intermittent in its nature and so I cannot reliably trigger > this issue, other than through sheer repetition, so any ideas as to what > could be causing this issue would be gratefully received, > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Video Key-Frame Request using RTCP FIR or SIP INFO message
Hi Muhammad Can you comment if initially endpoints are receiving what is necessary to start sending Media at signaling level. For example: Successful ICE and SRTP-SDES/DTLS negotiation. I see two issues here: a) Establish a successful call b) Once call is established how to deal with packet loss. Check this paper: http://static.googleusercontent.com/media/research.google.com/sv//pubs/archive/41611.pdf >From your email: "Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO message and issue a key-frame in RTP video stream in response to this SIP request?" WebRTC in the browser depends on a upper transport layer in your case a SIP stack. Example: sipml5, sip.js, etc. Hence you need to modify that part there; so your signaling stack should interact with the Browser Media Engine upon recieving SIP INFO. Questions: 1. I would suggest to start a conversation in discuss-webrtc in Google Groups. -Which SIP stack are you using on the WebRTC client side? -Can you upload logs from WebRTC client and SIP client. (WebRTC/SIP/SIP stack) -Topology and browser version -Codec: VP8/H.264. This will help us to understand how media is handled. If you do a packet capture can you still see Browser sending Video to SIP Client after those initial 5-7 seconds. (Check Webrtc logs/packet capture) Some details about WebRTC handling packet loss. https://groups.google.com/forum/#!topic/discuss-webrtc/0ZbxO05a9Zk HTH Thanks -G On Thu, Jan 29, 2015 at 2:56 PM, Muhammad Shahzad wrote: > Hi, > > This may be a bit out of focus topic for this forum but i am posting it > here anyway with hope that some guru would shed some light on it and point > me to right direction. > > The problem is that i want to establish video call between a webrtc and a > sip client using kamailio (for signalling) and RTPEngine (for media relay). > Both signalling and the audio stream seems to work perfectly fine The > remote video on webrtc client side (i.e. video stream from sip client) > takes about 20-30 seconds to establish but once it starts it works fine. > However, the remote video on sip client side (i.e. video stream from webrtc > client) starts almost immediately (within 3-5 seconds) but it gets stuck > after 1 or 2 seconds, then it goes blank after about 30 seconds. > > After a long discussion with sip client developer, we now understand the > fact that sip client sends a request for so called key-frame, which is > ignored by webrtc client. This request is sent through both RTCP stream and > SIP INFO message. > > The SIP INFO message seems to be pointless as media is internally managed > by chrome/firefox and these browsers don't give us such sophisticated > access and control over media streams. Please let me know if this > assumption is wrong. > > For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP > packet type" error message in RTPEngine logs (not sure if it drops this > packet or relay it anyway). > > Does anyone has any idea on how can we either, > > 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO > message and issue a key-frame in RTP video stream in response to this SIP > request? > > OR > > 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video > stream on webrtc client's behalf? > > If there is any other solution to this, please feel free to share. > > > Thank you. > > > > ___ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
