[OpenSIPS-Users] webrtc does not work

[Mikhail Sidorov]

Hi

I am trying to use opensips 2.2.3 + freeswitch + sip.js and I found that
loose_relay could not find proper websocket connection to send BYE if
client re-register during call.
Is there any way to handle this?

Steps to reproduce:
1. Connect 2 web clients using sip.js to opensips
2. Initiate the call
3. Wait for about a minute for client A to re-register
4. Let client B hangup the call.
5. Client A never receive hangup

logs:

Jun 21 09:04:26 s /usr/sbin/opensips[10054]: INFO:core:probe_max_sock_buff:
using snd buffer of 416 kb
Jun 21 09:04:26 s /usr/sbin/opensips[10054]: INFO:core:init_sock_keepalive:
TCP keepalive enabled on socket 23
Jun 21 09:04:26 s /usr/sbin/opensips[10054]:
ERROR:core:tcp_connect_blocking: timeout 99219 ms elapsed from 100000 s
Jun 21 09:04:26 s /usr/sbin/opensips[10054]:
ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
Jun 21 09:04:26 s /usr/sbin/opensips[10054]: ERROR:proto_wss:ws_connect:
connect failed
Jun 21 09:04:26 s /usr/sbin/opensips[10054]:
ERROR:proto_wss:proto_wss_send: connect failed
Jun 21 09:04:26 s /usr/sbin/opensips[10054]: ERROR:tm:msg_send: send() for
proto 6 failed
Jun 21 09:04:26 s /usr/sbin/opensips[10054]: ERROR:tm:t_forward_nonack:
sending request failed
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20170621/6b6d1111/attachment.html>