Hello,
I am using KSR.sdpops.keep_codecs_by_name to remove some codecs.
When I make a call between two webrtc clients and I keep only VP8 video
codec, the browser rejects the call with 488 Not Acceptable Here.
I notice the attributes a=rtcp-fb: of codecs removed are kept.
I removed that attributes with:
KSR.textops.replace_body_atonce("a=rtcp-fb:1.*\n", "");
KSR.textops.replace_body_atonce("a=rtcp-fb:98.*\n", "");
And the call worked.
There is another way to remove these attributes related to codec name?
Or is it possible to get the ids of the codecs with dynamic payload type
numbers from SDP?
Thanks for your attention.
Best regards,
Jose Lopes
Hello,
I encounter unexpected behavior.
The call flow is:
UAC --> Kamailio + Rtpengine (full ipv4) --> Asterisk (full IPV4).
I configured the uac to use IPV4 but the SIP INVITES include IPV6 in
headers and sdp messages. I think this is a bug on the UAC side.
Example:
1. UAC INVITE
*INVITE* sip:xxx@sip.xxx.com SIP/2.0 Via: SIP/2.0/TLS
[2a04:cec0:101a:ebe2:9949:d81c:23a6:ae10]:42580;branch=*z9hG4bK.I8U1RQDbW*
;rport
.....
.....
v=0
o=59130c8f7268a_2 2922 1557 IN IP6 2a04:cec0:101a:ebe2:9949:d81c:23a6:ae10
s=Talk
c=IN IP6 2a04:cec0:101a:ebe2:9949:d81c:23a6:ae10
2. We sent 200 OK response like :
Via: SIP/2.0/TLS
[2a04:cec0:101a:ebe2:9949:d81c:23a6:ae10]:42580;received=80.214.78.1;branch=
*z9hG4bK.I8U1RQDbW*;rport=1364
v=0
o=x 887503534 887503534 IN IP4 51.159.2.164
s=x
c=IN IP4 51.159.2.164
3. We never receive the ACK message.
Is there a Kamailio or Rtpengine solution to solve the problem?
Any help will be very appreciate.
Abdoul OSSENI
https://www.africallshop.com/
Hi, when I try to call Kamailio -> Teams.. this one always respònd to me
with a 404 Q.850 cause=1 (Unallocated number).
Could be this problem from Teams?
Has anyone ever encountered this problem?
Thanks
Hi, is it possible change ACK?
I have connected Asterisk to Kamailio like extension, with auth and secret.
I can redirect tha call from kamailio to my Asterisk but always in ACK, in
Asterisk, I received s@..... and in TO number@... is it possible send in
ACk the dialed number, like in TO??
Thanks
Hi,
I'm trying to set up a Kamailio server as a proxy to forward SIP messages
between two trunks, where I manipulate only the INVITE in the following
manner:
incoming call:
INVITE +4969123456@server1
outgoing call:
round robin to
INVITE user@4969123456-sip-server2a
INVITE user@4969123456-sip-server2b
INVITE user@4969123456-sip-server2c
Can anyone point me in the right direction? Both SIP trunks are IP
registration only. Also, I am aware how strange this is, but the upstream
service has decided to interpret SIP in a novel way.
Cheers,
Chris
Hi all,
I have configured Kamailio for WebSockets following this guide as an
example:
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
With sip.js and jssip I'm able to initiate a call from WebRTC to SIP and
establish a call successfully.
The issue arises when I try to receive a call from a SIP device. In this
case the call establishes but there is no audio in either direction.
I *think* the issue is with RTP Engine and I've raised a bug there, but I'm
not sure why it is misbehaving
https://github.com/sipwise/rtpengine/issues/983. There are some logs from
RTP engine posted here.
The sip device communicates with Kamailio over UDP / RTP, nothing is
encrypted.
I would appreciate any guidance.
Thanks in advance,
C
Hi,
Trying to use early media with prack support and rtpengine without success.
INVITE(no SDP)-->
183(SDP)<--
PRACK(SDP)-->
200ok(no SDP)<--
ACK(no SDP)-->
Looking at the code it does not seem to allow for PRACK in the methods it supports.
In the rtpengine_manage code...
if (!(method==METHOD_INVITE || method==METHOD_ACK || method==METHOD_CANCEL
|| method==METHOD_BYE || method==METHOD_UPDATE))
return -1;
In the rtpengine_answer code...
rtpengine_answer1_f(struct sip_msg *msg, char *str1, char *str2)
{
if (msg->first_line.type == SIP_REQUEST)
if (msg->first_line.u.request.method_value != METHOD_ACK)
return -1;
return rtpengine_rtpp_set_wrap_fparam(msg, rtpengine_answer_wrap, str1, 2, OP_ANSWER);
}
Returns -1 no matter what function you use.
I have checked code all the way up to 5.3 and on the master.
I may be looking at this wrong and it is supported, but it doesn't look like it.
Could you please advise if this is a oversight or if there is a specific reason it is not included.
Thanks
Chris
Chris Martineau | Senior Telephony Engineer
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Hello everyone!
I'm having an issue and was wondering if anyone could point me in the right
direction. I have a kamailio server that is crashing several times a day. I
see the following message in the logs:
>
> May 20 21:00:02 ip-10-192-11-197 /usr/sbin/kamailio[14705]: CRITICAL:
> <core> [core/pass_fd.c:277]: receive_fd(): EOF on 19
Then almost instantly the service restarts:
> May 20 21:00:02 ip-10-192-11-197 /usr/sbin/kamailio[14687]: ALERT: <core>
> [main.c:766]: handle_sigs(): child process 14689 exited by a signal 6
May 20 21:00:02 ip-10-192-11-197 /usr/sbin/kamailio[14687]: ALERT: <core>
> [main.c:769]: handle_sigs(): core was generated
> May 20 21:00:02 ip-10-192-11-197 /usr/sbin/kamailio[14687]: INFO: <core>
> [main.c:792]: handle_sigs(): terminating due to SIGCHLD
> May 20 21:00:02 ip-10-192-11-197 /usr/sbin/kamailio[14705]: INFO: <core>
> [main.c:847]: sig_usr(): signal 15 received
This is a dump of the log when this happens: https://pastebin.com/KAT9n0x9
The core dump doesn't make much sense to me, but this is it:
https://pastebin.com/9rVZjsEn
Any ideas on what may be causing this? Thanks in advance for reading!
--
Jacob Greene
*Voice Engineer*
Hello all,
I need to have two separate kamailio instances' dispatcher modules make the
same decisions when using algorithm 7 (hash over pvar). What do I need to
do to ensure this?
Note that for design reasons, the two instances cannot share a dispatcher
table from db. If I ensure the "setid" group used for algo 7 in the
respective cases contain the same group of hosts, is it enough? Do other
things matter, such as the ordering of the group members in the table (i.e.
different AUTO INCREMENT ids?). Does the setid need to be the same number?
Do I need to ensure the 'destination' values are identical (i.e. not using
IPs for dispatcher table A and hostnames for dispatcher table B)?
If someone knows what the criteria is for matching a hashed pvar to a
member of a dispatcher setid and how this can be made deterministic, I
would be grateful. Thanks!
BR,
George