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
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
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
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
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@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@67e046bmmlbt.invalid and invite gets routed properly to that uri.
-- juha
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
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@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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Brad Johns -
Juha Heinanen