[SR-Users] SIP client TCP behaviour

Hello, I don't know a lot about SIP with TCP, so I thought I'd ask for some opinions here. I've been testing Kamailio with TCP against the Android SIP client (4.4.2) and have found, to my annoyance, that it sends a RURI with a transport attribute in its INVITEs: INVITE sip:14045551212@sip.evaristesys.com;transport=tcp My upstream gateways are all UDP, so this causes Kamailio to attempt to contact them all via TCP (and fail). I know how to bend the transport with Kamailio, and I can strip this attribute. My question is more methodological: is this "correct" for a client to do? In my opinion, the answer is no. The client should not be so presumptuous. It should specify the transport in its Contact, to indicate how it wants to be reached, and leave other decisions about transport to the UAS. But is there something I am perhaps missing? Thanks, -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/