[VoiceOps] TCP Signaling for SIP Signaling

[Colton Conor]

I know UDP seems to be the gold standard for SIP, and is in use by most
service providers that are offering hosted voice today. My question is why
not use TCP instead of UDP for SIP signaling?

Overall with small business clients we run into firewalls with SIP ALGs,
short UDP session time out limits, and all sorts of connectivity issues
with UDP. Some small business routers and modems have built in SIP ALGs
that can't be disabled at all. The second we switch to TCP for signaling
most of the issues go away for our hosted voice customers. Overall TCP just
always seems to work, and UPD depends on the situation of the network. TCP
is better for battery consumption on mobile sip applications as well.

With more providers switching to encryption using TLS which uses TCP, is
there any need for us UDP for signaling anymore? Assuming most IP phones
from Polycom, Yealink, and Cisco support TCP why not use it? Is it more
resouce intensive on the SBCs?

What about on the media side? Does the RTP use UDP or TCP? If it uses UDP
can TCP be used? What about for encryption like SRTP? Is SRTP TCP or UDP?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/voiceops/attachments/20170716/01da3c5c/attachment.html>