Hi,
I installed and configured the latest Kamalio and all the features works
except PRESENCE
Also it works with NAT clients with RTPProxy.
When I register with Jitsi or QuteCom I get
5(10515) ERROR: presence [publish.c:388]: No E-Tag and no body found
6(10517) INFO: presence [notify.c:1601]: NOTIFY sip:1001@sip.abc.com via
sip:1001@124.43.201.156:5060 on behalf of sip:1000@sip.abc.com for event
presence
6(10517) ERROR: presence [publish.c:388]: No E-Tag and no body found
ERROR: presence [publish.c:388]: No E-Tag and no body found
Please advice me how to fix this.
Best Regards,
Roy.
Hi,
As someone asked in
http://lists.sip-router.org/pipermail/sr-users/2010-December/066792.html about
TCP overload, Daniel answerd that
"if you have some lengthily operations that have to be applied to your sip
traffic and you get congestion because of that, the solution is to use a
kamailio as load balancer in front of a farm of proxy doing the actual
processing"
How to use Kamailio as load balancer ?
--
Khoa Pham
HCMC University of Science
Faculty of Information Technology
Hello all,
In the documentation of the t_relay_cancel() (TM module) there is an
example that reads:
if (method == CANCEL) {
if (!t_relay_cancel()) { # implicit drop if relaying was successful,
# nothing to do
# corresponding INVITE transaction found but error occurred
sl_reply("500", "Internal Server Error");
drop;
}
# bad luck, corresponding INVITE transaction is missing,
# do the same as for INVITEs
}
What bothers me is the phrase #do the same as or INVITEs, because in RFC(
http://tools.ietf.org/html/rfc3261#section-16.10 ) it says:
If a response context is not found, the element does not have any
knowledge of the request to apply the CANCEL to. It MUST *statelessly*
forward the CANCEL request (it may have statelessly forwarded the
associated request previously).
So aren't we supposed to immediately statelessly forward the CANCEL if
t_relay_cancel() did not find the INVITE transaction, instead of doing
the same as INVITEs, which could be t_relay() (not stateless)?. Like
below:
if (method == CANCEL) {
if (!t_relay_cancel()) { # implicit drop if relaying was successful,
# nothing to do
# corresponding INVITE transaction found but error occurred
sl_reply("500", "Internal Server Error");
drop;
}
# bad luck, corresponding INVITE transaction is missing,
forward();
}
Am I correct or am i missing something?
Thank you in advance,
Bill
my presence server tries to handle subscribe to
sip:jh+presence@test.tutpro.com. that service uri is stored in
rls-services document of user sip:jh@test.tutpro.com, but for some
reason k does not find the service uri.
here is some debug:
May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: Handling SUBSCRIBE <sip:jh+presence@test.tutpro.com>
May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:155]: searching document for user sip:jh@test.tutpro.com
May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls
[subscribe.c:240]: rls_services document:#012<?xml version='1.0'
encoding='UTF-8'?>#012<rls-services
xmlns="urn:ietf:params:xml:ns:rls-services"><service
uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service><service uri="sip%3Ajh%2Bdialog%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_dialog_rls%22%5D</resource-list><packages><package>dialog</package></packages></service></rls-services>
May 1 19:08:43 siika /usr/sbin/pres-serv[27355]: INFO: rls [subscribe.c:253]: service uri sip:jh+presence@test.tutpro.com not found in rl document for user sip:jh@test.tutpro.com
any idea why the service uri is not found even when it appears to be in
the document:
<service uri="sip%3Ajh%2Bpresence%40test.tutpro.com"><resource-list>http%3A//192.98.102.10%3A8080/xcap-root/resource-lists/users/sip%3Ajh%40test.tutpro.com/index/%7E%7E/resource-lists/list%5B%40name%3D%22sipsimple_presence_rls%22%5D</resource-list><packages><package>presence</package></packages></service>
could the escaped chars explain the failure or the fact that
resource-list is an uri?
-- juha
Hi All,
I need forking a call to multiple destinations in paralel on different
network segments, requiring different rtpproxy parameters. Reading the
rtpproxy module documentation I discovered that is possible by setting the
"b" parameter on rtpproxy_manage(), rtpproxy_offer(), rtpproxy_answer() and
rtpproxy_destroy() functions, but I'm with some doubts...
I can't understanding how set the extra_id_pv parameter correctly to use
when the "b" parameter is used.
The "b" parameter is supported on rtpproxy version 1.2.1?
Best Regards
Hi,
Currently, I'm using TLS and it works fine. But eventually, TLS is just
used to agree upon a secret key. And both client and Kamailio use that
secret key to encrypt message.
In my solution, I have a fixed secret key. And client encrypt SIP message
with that secret key.
What code should I insert in Kamalio to decrypt it?
--
Khoa Pham
HCMC University of Science
Faculty of Information Technology
Last I looked (around February) there were some things in Kamailio
presence that aren't in OpenSIPS.
For example, the separate notifier processes in both presence and rls (I
believe one may have been ported across now by Saul - but not both).
There are also extensions relating to the built-in XCAP server
(particularly in presence_xml and rls) that are in Kamailio and not
OpenSIPS. I know that there are some specific RLS fixes and features in
Kamailio that only work when you use the built-in XCAP server. I also
made some enhancements to the way pidf-manipulation is handled in Kamailio
that I do not think have been ported to OpenSIPS.
Basically, while OpenSIPS presence may be more advanced in some ways I am
quite sure it is less developed in some others. Having spent a lot of
time over the course of around 18 months doing work in Kamailio presence I
would strongly resist any attempts to junk what is currently there because
I know that I need a lot of the stuff that is in the Kamailio
implementation. I don't currently have the time to develop it further at
the moment either.
I think the "junk them and replace with opensips" approach would be a
disaster.
Regards,
Peter
> looks like kamailio presence and especially rls is lagging behind in
> development as compared to opensips. it properly handles the escapes
> that i had trouble with, support external references in presence
> rules, xcal-diff, etc.
>
> what should we do about kamailio presence modules? junk them and
> replace with opensips ones or implement the missing stuff ourselves?
> i personally don't have resources for the latter.
>
> -- juha
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
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>
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
Thanks for reply,
I checked log messages at callee (IPv6 client - Linphone) and result was little surprising for me.
This IPv6 client as Callee didnt receive INVITE messages (I thought that yes because of Wireshark at SIP server).
So it is my bad somewhere in my local network.
But it is little interesting, that in other direction (from IPv6 caller to IPv4 callee) all Reply messages are received at IPv6 client and RTP/UDP packets go in both directions.
I checked Iptables and Firewall at all nodes, but nothing (they all are in the same network) and I dont know what filteres messages to IPv6 callee client.
btw. the proxy server in the middle is SIP Protocol gateway, which translates messages between IPv4 and IPv6 and packets at this node are OK.
thanks for help
Jakub Hrabovsky
