[asterisk-users] SIP via TCP - new TCP session per call or use same session for multiple calls?

[Greg Troxel]

"Joshua C. Colp" <jcolp at sangoma.com> writes:

>> I am curious if the "reuse registration TCP connection" is required by
>> standards or if it is merely obviously good practice.
>>
>> I have had this problem too with asterisk 16.5.0
>>
>> This is not the first recommendation I have seen to use kamailio as a
>> proxy for asterisk, for these sorts of issues as well as clients that
>> change addresses.  Unfortunately the "jsr pc, set_up_kamailo" subroutine
>> call is still executing so I can't say if things work right then...
>>
>
> There is a specification for doing it, but it's not required by the main
> SIP RFC. In fact the main one states that you're supposed to establish an
> outgoing connection to the address in the Contact header. In practice,
> though, this is futile as generally NAT Is in use so you can't connect back
> and thus you reuse the connection. The chan_sip module should do this
> automatically, while chan_pjsip will do this if "rewrite_contact" is set to
> yes.

Thanks.  I read the docs for pjsip (and the book) and failed to grasp
that.  It seems that this should be the default behavior.