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

Daniel-Constantin Mierla miconda at gmail.com
Wed Jun 30 12:24:51 CEST 2021

On 30.06.21 09:28, Juha Heinanen wrote:
> Shahid Hussain writes:
>> Would like to know what is the recommended solution for this problem using
>> alias or is it a limitation of using alias?
> Maybe a limitation.  Try with SIP User Agents that support gruu and thus
> identify themselves using sip.instance.

Relying on GRUU would be ok, but most of the UAs I had to deal with lack
its implementation.

What I have seen as an alternative (e.g., linphone, iirc) is sending
re-INVITE with the new contact address and then a new alias will be
generated as well. It is not enough for the UA to send a new REGISTER,
that is for receiving new calls. In making calls is not restricted to
registered devices, therefore when an UA changes is local
states/coordinates, it has to send a re-INVITE. Contact can be changed
within dialog (also, just for sake of completion, dialog module should
save new Contact values), only Record-Route/Route headers are fixed for
active calls (as far as I remember right now).


Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda

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