[SR-Users] Modern SIP Trunking, Route on To: or INVITE Headers?

JR Richardson jmr.richardson at gmail.com
Tue May 21 15:32:20 CEST 2019


Hey Folks,

I could use some feedback to see if my mind is right on this topic.I'm
in a discussion with a software Vendor (respectfully unnamed) with
terminal software routing on To: header info. It's my contention any
modern SIP software should use INVITE header to retrieve DID info and
route from there.

My reasoning is due to the To: header should contain the original
intended DID, Extension, Name, AoR or whatever else the original SIP
transaction wanted to contact, but being in a multi-carrier,
multi-hop, call forwarding environment, the INVITE header contains the
hop-to-hop true intention of where the call should route at any given
transaction. So in the case of terminal software the To: header could
likely be irrelevant where as the INVITE header is accurate.

My most recent response from the Vendor suggest I put a SIP Proxy in
front of the terminal software and transform the To: header to
whatever I need for this application. Although I can do this, I'm
concerned with breaking RFC for CANCEL and BYE transaction sent back
from the terminal software that may not be recognized by upstream
origination carriers.

Plus modifying To: header is widely frowned upon by most folks.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses
Chasing the Azeotrope



More information about the sr-users mailing list