Hello,
I am using rtpengine to encrypt and decrypt DTLS and SDES audio streams for my calls. I have a set of rtpengine instances running, configured using a database, such that it is easy to scale up on demand. This works well for initiating a call and balancing load across multiple instances but I want to be able to recover from rtpengine failure.
I have looked at configuring rtpengine with a redis service but do not believe that to be helpful in this case as im using encrypted media and because the configuration of rtpengine with redis has a lot of overheads, limits scaling and requires non standard configuration of a redis cluster.
I would like to be able to re-anchor media streams instead of relying on rtpengine to synchronise itself across a redis cluster. For this to work, i believe i would have to use the uac module to send a reINVITE and I speculate that i would need to use an xhttp event to trigger this remotely when a failure is detected.
I am unclear, though, how to get to a specific call within kamailio from the xhttp route in order to do anything to it, if i even need to, and i am nuclear how to make the rtpengine module select a new node for sdp for the reinvite?
How can i achieve my goal?
Perhaps it would be possible to modify the rtpengine module itself to support generating a reinvite and selecting a new node on receipt of an rpc request?
Hoping someone can help me find a way forward
With every blessing,
Daniel
Hi,
I am using kamailio 5.8.3 as an IMS-PCSCF. On handling SIP INVITE (and
response), it sends Rx AAR to PCRF for bearer establishment.
For mobile-terminated call, it was found that the call was dropped by
handset within several seconds after ringing for some handsets.
The corresponding pcap capture is attached for reference.
From the pcap capture, it was found that in the Rx AAR request sent (packet
11), in the Media-Sub-Component Flow-Description,
the "permit out from ip/port" and "permit in to ip/port" were set to
202.0.169.182 port 24428 instead of 10.51.0.15 port 31124.
202.0.169.182 port 24428 is the one in the incoming INVITE sdp from
S-CSCF (packet 2);
10.51.0.15 port 31124 is the one in the processed INVITE sdp sent to UE
(packet 5);
Also, the sdp data of "offer" inside AAR was set to the sdp in the incoming
INVITE from S-CSCF instead of the sdp in the processed INVITE sent to UE.
Is that correct? Please advice.
Thank you.
Regards,
Hong
How can I set the destination URI for an INVITE to be a
websocket-secure destination? Is it possible?
Summary
I've a proxy with tcp_connection_match=1, but websocket URIs always
have transport=ws (never transport=wss) in them, so relaying a call to
a WSS connection always fails.
I tested running kamailio 6.0.0-dev2 compiled from a commit made this
week. This proxy server uses nathelper rather than outbound module.
Detail
We know that "transport=ws" is used for both WS and WSS. I've a proxy
server that receives an INVITE for a WSS destination, and this proxy
supports both WS and WSS.
This proxy server must have core parameter tcp_connection_match=1 set,
and this leads the t_relay() to fail.
When an INVITE comes, these are the steps.
- The URI is something like
sip:user@anonymous.invalid;alias=198.51.100.10~52833~6;transport=ws.
- First handle_ruri_alias() removes the alias (which has ~6 in it, for
wss) and sets the $du to something like
sip:198.51.100.10:52833;transport=ws.
- Then loose_route_preloaded() processes the Route header fields and
forces the outbound socket to the TLS websocket one.
- Then t_relay() fails to relay the INVITE and responds with 477 or 500.
If, however, there's a non-TLS websocket connection open to the proxy,
the INVITE would be erroneously relayed over that (using the wrong
kamailio-side TCP port).
I can go deeper with testing if required. I wonder whether this is a bug.
James
Hi,
I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size).
In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line,
$var(used) = 1 - ($stat(free_size) / $stat(total_size));
The trace log said that $stat(total_size) was returning 0, and got a log like this
ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0
Any idea on this? Please advise.
Thank you.
Regards,
Hong
Hi,
I am using kamailio 5.8.3. I just found that strange result is returned from variable $stat(total_size).
In the example script under "misc/examples/ims/pcscf/kamailio.cfg", there is a line,
$var(used) = 1 - ($stat(free_size) / $stat(total_size));
The trace log said that $stat(total_size) was returning 0, and got a log like this
ERROR: <core> [core/rvalue.c:1434]: long_longop2(): rv div by 0
Any idea on this? Please advise.
Thank you.
Regards,
Hong
Hello,
We try to introduce a small websocket sidecar to our Kamailio servers, which runs on the same host. It connects to the regular TCP socket of our Kamailios, and in the XHTTP request route, we do just this:
if path == "/sidecar":
KSR.websocket.handle_handshake()
return 1
On every new connect, Kamailio logs a parse error for the reply it generates.
Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/parse_fline.c:272]: parse_first_line(): parse_first_line: bad message (offset: 22)
Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/parser/msg_parser.c:790]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols
Sia: SIP/2.0/TCP 100.74.11.1:46422
Sec-WebSocket-Protocol: sip
Upgrade: websocket
Connection: upgrade
Sec-WebSocket-Accept: 60dPbzcRW6up7z92kDlxkqL2WlY=
Content-Length: 0
>
Sep 11 15:42:35 sipproxy /usr/sbin/kamailio[552]: ERROR: <core> [core/msg_translator.c:3402]: build_sip_msg_from_buf(): parsing failed
I saw this error in other questions from a while ago, but without a solution. What could be the reason for this?
Thanks in advance,
Sebastian
Hello guys,
i'm seeing something weird, and i'm working if you can let me know.
I have a kamailio in AWS with a private IP listening on several sockets:
Listening on
udp: 10.1.2.36:5070
udp: 10.1.2.36:5080
udp: 10.1.2.36:5160 advertise 4.3.2.1:5160
udp: 0.0.0.0:5066
tls: 10.1.2.36:443 advertise sip.something.com:443
tls: 10.1.2.36:444 advertise sip.something.com:444
tls: 10.1.2.36:5061
When forwarding a udp invite received on 10.1.2.36:5080 to a public ip
provider say on 8.8.8.8:5060, i'm forcing the outgoing socket with
force_socket via 10.1.2.36:5160. But the outgoing invite does NOT use 5160,
it uses some random port...
Anybody knows why this might be?
my problem is, that call goes to freeswitch... call is setup properly and
connects fine. But 15 minutes later the end provider sends back a reINVITE,
which freeswitch then sends TO THE RANDOM PORT kamailio used to send the
INVITE... but by this time kamailio doesn't seem to even see the packet...
help is greatly appreciated!
David
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
Hi,
I just tried to send numbers of SIP register to kamailio (release 5.8.2). It is reported that there are some memory leakage in shared memory after the expiry of all the registration sessions. After drill down the code. function save_pending(...) is called. (in ims_registrar_pcscf/save.c). Inside that function, it seems that "sec_verify_params" is not deallocated upon the function completes.
Findings:
- sec_verify_params uses the result from function cscf_get_security_verify(...)
- Function cscf_get_security_verify(...) returns the result of function parse_sec_agree(...)
- Function parse_sec_agree(...) returns "params" where "shm_malloc(...)" is called
It seems that the memory pointed by 'sec_verify_params' is not further referenced by others. Should shm_free(...) be called when leaving function save_pending(...) in order to free the unused memory of 'sec_verify_params'?
Please advice.
Thank you.
Regards,
Hong
Hello,
I was wondering if icc or suncc compilers are still used in the
community, and by that if it still worth trying to support them via
Makefiles and compile time flags (defines). Personally I haven't had to
deal with them for more than 10-15 years, I have no access with a system
offering them. Furthermore, I am not even sure if the code compiles with
them.
Maybe someone (or many) in the community use one or the other and
confirm that they still worth to be kept.
If not, maybe we can consider them for removing, it can be a good task
to be done during Kamailio development meeting planned in Dusseldorf,
Germany during November.
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
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