[VoiceOps] TCP Signaling for SIP Signaling

[Peter E]

The SIP protocol already has some built in reliability techniques built in (timers, retransmission) for which TCP is usually used. Yes, TCP is a bit more resource intensive due to the TCP overhead. But the biggest reason I prefer UDP is for failover/redundancy. In the event of a system failure/failover, UDP will be in-disrupted but TCP will. Your TLS argument is valid, however.

RTP will always be UDP. Think about it... TCP will retransmit when packers are lost, but in real time communication there is no need to retransmit. While packet loss is problematic, a retransmission of lost packets would be unexpected and cause further quality issues. 



On Jul 16, 2017, at 22:28, Colton Conor <colton.conor at gmail.com> wrote:

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?


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