[SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

[Shahid Hussain]

Hi,
Websocket module documentation has a code reference to use aliases for SIP
routing. However, aliases will not work in the following setup and
situation.
1. Kamailio is configured with active and standby node
2. Ping is implemented from webclient and kamailio responds with pong.
3. Two clients ClinetA and ClientB registered themselves to Kamailio.
4. After SIP negotiation (INVITE-200OK) each client learnt about below
aliases.
Alias of ClientA:
alias=172.27.6.98~58336
Alias of ClientB:
alias=172.27.6.98~58342

So normally if ClientB wants to send SIP message to ClientA, SIP URI from
ClientB looks like below
ACK sip:v9d0gtl6 at q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws
SIP/2.0
 4. Call is in a connected state.

Following is the issue.
i.  Switchover (or network lost or reboot) at Kamailio happened
ii.  Due to ping pong both the clients detected network loss individually
and re-registered themselves.
iii.  Aliases of both the clients got changed.
New Alias of ClientA:
alias=172.27.6.98~ 58346

New Alias of ClientB:
alias=172.27.6.98~ 58348

iv.   But ClientB doesn’t know that ClientA also re-registered and
ClientA’s alias got changed and vice-versa
v.    Because of this ClientB still sends BYE with Initial alias
BYE sip:v9d0gtl6 at q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws
SIP/2.0

Would like to know what is the recommended solution for this problem using
alias or is it a limitation of using alias?

Thanks & Regards,
Shahid
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