Hi,
I have a WebRTC softphone based on JsSIP and kamailio 4.4.7. Generally it
works well but sometimes websocket connections interrupt and reconnect
immediately (especially running behind proxies). If the interruption occurs
during an active call, kamailio tries to send BYE message to softphone over
closed websocket.
gruu_enabled flag is active on registrar module and JsSIP sends proper
sip.instance value. So after ws reconnects, registered contact details do
not change on kamailio except Expires value.
Is it possible to send BYE message via contact's last active websocket
after reconnect?
Jan 29 14:35:20 registrar-kenan /usr/local/sbin/kamailio[30357]: WARNING:
<script>: WebSocket connection from 195.142.112.66:49086 has closed
Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: WARNING:
<core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0)
for WebSocket could not be found
Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR:
<core> [msg_translator.c:1979]: build_req_buf_from_sip_req(): could not
create Via header
Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: tm
[t_fwd.c:462]: prepare_new_uac(): could not build request
Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: tm
[t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add
branches
Jan 29 14:35:24 registrar-kenan /usr/local/sbin/kamailio[30323]: ERROR: sl
[sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error
(2/SL)
Regards,
Kenan