[SR-Users] SIP client TCP behaviour

[Alex Balashov]

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 at 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/