[SR-Users] SIP client TCP behaviour
abalashov at evaristesys.com
Mon Apr 7 17:53:24 CEST 2014
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?
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Decatur, GA 30030
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
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