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. 

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