28 Mar
2013
28 Mar
'13
11:23 p.m.
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.
However, after registration, the users can't place an audio call. I see no
ringing on the remote browser. I don't know how to debug this further to
find out what the problem is. Can anyone help with clues or debug? In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE. However, nothing
indicating a call.
With this JsSIP, I can do chat through Kamailio SIP over WebSockets.
With this Kamailio server, SIP User Agent Clients work just fine to
register and place SIP call with audio.
It's just that WebRTC audio calls don't work with JsSIP sample application
with Kamailio 4.0 websocket module.
Kamailio websocket configuration borrowed from:
https://gist.github.com/jesusprubio/4066845
Any help debugging this appreciated.
Brad
29 Mar
29 Mar
9:52 a.m.
Brad Johns writes:
However, after registration, the users can't place
an audio call. I see no
ringing on the remote browser. I don't know how to debug this further to
find out what the problem is. Can anyone help with clues or debug? In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE. However, nothing
indicating a call.
i just tried tryit app with chrome 26 and calls worked for me. there
was considerable delay (about 10 sec) before the called browser showed
the call coming in, but once it did, i was able to answer the call and
see video flying both ways. i didn't hear any audio, which may be due
to audio settings of the browser.
Any help debugging this appreciated.
use wireshark to see what is going on on the wire.
i noticed that tryit app uses normal sip tcp port 5060. i didn't even
have websocket.so loaded in my proxy when i did the test. perhaps it
falls back to sip over tcp if sip over websocket transport is not
supported?
-- juha
9:59 a.m.
Juha Heinanen writes:
i noticed that tryit app uses normal sip tcp port
5060. i didn't even
have websocket.so loaded in my proxy when i did the test. perhaps it
falls back to sip over tcp if sip over websocket transport is not
supported?
by bad, i had not reconfigured WS URI and therefore the default proxy
was used as outbound proxy.
i'll try again.
-- juha
10:10 a.m.
Juha Heinanen writes:
by bad, i had not reconfigured WS URI and therefore
the default proxy
was used as outbound proxy.
i'll try again.
now that tied with my own proxy where i had activated websocket
transport, tryit app worked fine. i was even able to hear voice and see
video.
does anyone know how to make wireshark to properly display sip over
websocket transport?
-- juha
10:21 a.m.
when tryit app sends invite to my proxy, i get this to syslog:
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: INFO: INVITE <sip:bar@test.fi> by
<foo(a)test.fi> as <sip:foo@test.fi> from <192.98.102.131> is authorized
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: INFO: registrar [lookup.c:313]: instance is
<urn:uuid:cf0efb96-f864-45a6-8cd0-88b7b33a3289>
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: INFO: Routing INVITE to contact
<sip:fc8sa46v@tfq7u3ur8iqs.invalid;transport=ws;ov-ob=48f3da1b68>
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: : <core> [dns_cache.c:3374]: BUG:
sip_resolvehost: unknown proto 5
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: ERROR: tm [ut.h:314]: ERROR: uri2dst:
failed to resolve "tfq7u3ur8iqs.invalid" :bug - critical error (-13)
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: INFO: Routing INVITE to contact
<sip:et9oloai@67e046bmmlbt.invalid;transport=ws>
i don't know where the bug/error comes from, since i do not have any uri
in usrloc that would have host tfq7u3ur8iqs.invalid. i do have
et9oloai(a)67e046bmmlbt.invalid and invite gets routed properly to that uri.
-- juha
12:04 p.m.
Juha Heinanen writes:
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: :
<core> [dns_cache.c:3374]: BUG: sip_resolvehost: unknown proto 5
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: ERROR: tm [ut.h:314]: ERROR: uri2dst:
failed to resolve "tfq7u3ur8iqs.invalid" :bug - critical error (-13)
that bug/error disappeared after some time passed. perhaps something
was left in dns cache from previous registration even when it was
already removed from usrloc.
-- juha
10:24 p.m.
Very interesting. I am still not able to have an audio call complete. Can
I see your kamailio.cfg, under separate cover? Or sent to the list?
Do you have "method_filtering" in registrar params set to 0 or 1? I had it
set to 1 and by default the JsSIP tryit must not have been sending an Allow
of REGISTER. Once I set it to 0, my kamailio could process the invite.
Glad to hear it worked so easily for you. I must have some errors in my
configuration. I am trying to do MySQL for authentication and accounting,
but that shouldn't be an issue. The SIP user agent clients work just fine
with MySQL. Don't know why I'm having all of these problems with WS.
Brad
On Fri, Mar 29, 2013 at 5:04 AM, Juha Heinanen <jh(a)tutpro.com> wrote:
Juha Heinanen writes:
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]: :
<core>
[dns_cache.c:3374]: BUG: sip_resolvehost: unknown proto 5
Mar 29 10:04:13 sars /usr/sbin/sip-proxy[626]:
ERROR: tm [ut.h:314]:
ERROR: uri2dst: failed to resolve
"tfq7u3ur8iqs.invalid" :bug - critical
error (-13)
that bug/error disappeared after some time passed. perhaps something
was left in dns cache from previous registration even when it was
already removed from usrloc.
-- juha
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Juha Heinanen