Hello.
I’m having some problems using websocket to communicate a webRTC client
with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket
I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length.
When I made the capture on the other leg (the
Hello,
have you printed the message in syslog with $mb to see its content?
What is the content-lenght value sent by client and the one sent out by
kamailio.
Cheers,
Daniel
On 28/10/14 16:21, Ricardo Martinez wrote:
Hello.
I’m having some problems using websocket to communicate a webRTC
Hello,
the problem is that the callee is behind the nat but you don't handle
the 200ok properly in order to update the contact address. The contact
address stays like 192.168..., which cannot be routed by kamailio when
it appears in R-URI of ACK and BYE.
Be sure you do nat traversal logic for
Hello,
content-length checking is done by sanity_check() function from sanity
module, used in route[REQINIT]. I do not recall any changes to this
function for 4.2. Anyhow, you can remove the check on content-length if
you want, see the readme of the sanity module to get the value of
appropriate
Hello,
We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC
Proxy doing the encryption and NAT Traversal.
Everything works as expected, except that BYE Requests sent by the WebRTC
Client are not forwarded by Kamailio to Asterisk. We use record routing.
Instead Kamailio
Hello,
should be fixed in 4.2 -- the issue was introduced when changed the
build of refer to contain a contact header, as it was reported some UA
don't like it without the header.
Let me know if all works ok now.
Cheers,
Daniel
On 24/10/14 14:56, Daniel-Constantin Mierla wrote:
I don't recall
Hello,
I done the manipulation.
I still have some errors incorrect port 0 in reply from rtp proxy with a
moderate flow of calls.
I don't really know where to look at for solve this issue.
Any other ideas? Thank you.
Regards,
Igor.
-Message d'origine-
De : Igor Potjevlesch
Hello,
the BYE is coming with a rather strange R-URI. That should be taken from
INVITE contact. Also, apparently the INVITE comes from behind NAT, you
should use nat traversal logic to update the contact (e.g., add/set
alias parameter).
See default configuration file for nat traversal, same
Hi There,
since 4.2 we had the problem that kamailio did not start after a reboot.
A /etc/init.d/kamailio restart fixxed it. The problem is quite obvious:
kamailio-loglile = Kamailio can't connect to mysqlDB 13:33:44
Oct 29 13:33:44 kamailio-test /usr/sbin/kamailio[1063]: DEBUG: db_mysql
Hello Kamal:
This is something to do with user location in this case IMS LOCATION/ Presence
IMS_USRLOC_PCSCFhttp://kamailio.org/docs/modules/4.1.x/modules/ims_usrloc_pcscf.html
* IMS PCSCF usrloc module released
Hi Zaka
Thanks a lot.
I have these modules, not sure why failing.
I commented out that portion of code, and looks it is working now.
Best Regards
kamal
On Wed, Oct 29, 2014 at 6:21 PM, Zaka Ul Isam zaka...@albtelecom.al wrote:
Hello Kamal:
This is something to do with user location in
Hello. I use kamailio with last rtpengine and
I have 5-7 Seconds voice delay. This happened only for from webphone. But
it is not client issue as i see. Wireshark at client side shows that RTP
starts as soon I pick up call. So rtp leaves rtpengine and goes to the
destination with delay... I use
On 10/29/14 10:08, Yuriy Gorlichenko wrote:
Hello. I use kamailio with last rtpengine and
I have 5-7 Seconds voice delay. This happened only for from webphone.
But it is not client issue as i see. Wireshark at client side shows that
RTP starts as soon I pick up call. So rtp leaves rtpengine
Thank you for the fast reply,
I have enabled NAT Traversal like in the default config. The problem seems
to be that the cannot be assigned to any transaction.
2014-10-29 13:08 GMT+01:00 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
the BYE is coming with a rather strange R-URI. That
I narrowed down the problem to a comparison made in this function:
send_rtpp_command(struct rtpp_node *node, struct iovec *v, int vcnt) line 1618
inside the rtpproxy.c file (kamailio-4.2/kamailio/modules/rtpproxy/rtpproxy.c,
lines 1693, 1694 )
if (len = (v[0].iov_len - 1)
memcmp(buf,
The question is, how I can forward the BYE message back to asterisk?
2014-10-29 15:24 GMT+01:00 Marko Seidenglanz marko.seidengl...@gmail.com:
Thank you for the fast reply,
I have enabled NAT Traversal like in the default config. The problem seems
to be that the cannot be assigned to any
Readers,
I am having issues with passing the max-forward header to my freeswitch
service.
Here are the sip invites
Here is a call coming from a kazoo box though my carrier Into kamailio.
Kamailio shows it has Max-Forward after the sanitize check, but when it
reaches freeswitch no Max-Forward
Hello,
most probably Max-Forward matches this:
remove_hf_re(X-.*);
Iirc, the regexp is case insensitive. You should use:
remove_hf_re(^X-.*);
In this way you are sure you don't match any X- inside header name.
Cheers,
Daniel
On 29/10/14 16:29, Mike Dunton wrote:
Readers,
I am having
Hi Daniel,
I’ve just done a few quick tests after “git pull origin” upgrade.
It works when the 2 snom endpoints are registered over UDP transport with stun
enabled. Which is great thank you very much.
But I have come across a couple of cases that are not working for me yet:
FAIL CASE 1 : It
Hello,
I tried to install the snmpstats module following this process :
I installed the dependencies for snmpstats module, then i modified
modules.lst.
Then i made a make modules, make, and make install.
Everything is fine so far.
After that, i edited kamailio.cfg the loadmodule for snmstats.so
Hello. I use kamailio for calling to porvider. My providr seccefuully
registered from UAC module, but when I try to call through it? it back 401
Unauthorised. I send second try with Digest Auth header at INVITE and it
receive me 401 too...
I register this provider from asterisk and call
Hi
We are having an odd issue with our LCR. This started after we upgraded
from kamailio 4.1.0 to 4.1.6.
Here is the situation:
We populate our gw_uri_avp (stored in: $avp(i:709)) as follows:
16|1[gateway ip address]||5060||1|1.
The $ru before running next gateway shows: sip:[destination
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