[VoiceOps] TCP Signaling for SIP Signaling

[Colton Conor]

Pete,

That makes sense. Lets assume a user that was registered to the west coast
SBC using UDP. If the west coast was to fail and they were on an active
call, its not like the call would continue and bounce over to the East
coast right? The call would drop, and the customer would have to call back
right?

Assuming you have your phones register every 5 minutes or so how would that
differ then UDP vs TCP? Are you saying UDP would instantly re-register to
the east cost node, but TCP would wait 5 minutes to it re registers?

I have noticed in the Broadsoft device config files provided by Broadsoft,
that Broadsoft themselves recommends TCPpreferred: TCP is the preferred
transport, UDP is used if TCP fails. However, I have yet to see any
Broadsoft based providers make TCPpreferred as the standard for the Polycom
phones. Everyone seems to leave it on the default of DNSnaptr or UDP only,
and uses UDP signaling. The question is why aren't they following the
Broadsoft lab tested and recommended approach?





On Mon, Jul 17, 2017 at 8:32 AM, Pete Eisengrein <peeip989 at gmail.com> wrote:

> Perfect example. With an Acme SBC as a redundant pair, if (when) the
> primary fails and switches to the standby, all UDP immediately goes to the
> standby and is generally unnoticed by the end users. However, if the SIP is
> over TCP in that scenario, the switch still happens but the TCP session
> must re-establish itself to the secondary SBC and therefore is an outage
> for those users until they re-register. I suppose it is possible to share
> TCP session info, but I am not aware of any SBC's that do this any
> differently (though would love to hear from the group if one exists).
>
> Somewhat related, but not really: It's since been patched but Acme
> (Oracle) had a bug at one point where it was not releasing TCP sessions
> after they were gone and you'd end up using all available resources (TCP
> ports); if that happened you'd begin blocking new sessions and the only
> workaround was a forced switchover which, of course, then meant a forced
> outage for those users.
>
> -Pete
>
> On Mon, Jul 17, 2017 at 9:01 AM, Colton Conor <colton.conor at gmail.com>
> wrote:
>
>> Thanks for the replies. I am mainly talking about using TCP for SIP
>> signaling for access/customer side of the network only. I think trunk
>> connections to carriers will stay UDP only for a long time.
>>
>> Overwhelming it seems like using TCP for signaling doesn't seem to be a
>> bad thing, and preferred by many. Peter, I have a question about what you
>> mean by "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."
>>
>> Lets assume we are using Broadsoft with an Acme Packet SBCs, and have
>> redundancy having one on the West coast and one on the east cost.
>>
>> Using TCP for signaling, how would this be different than using UDP in a
>> fail over secenario? Assume the client is closer to the West cost node, and
>> the West coast node rebooted or shut down due to power failure.
>>
>>
>> Using UDP for RTP makes perfect sense. Sorry for asking the stupid
>> question about RTP.
>>
>> On Sun, Jul 16, 2017 at 9:59 PM, Peter E <peeip989 at gmail.com> wrote:
>>
>>> 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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>>> VoiceOps at voiceops.org
>>> https://puck.nether.net/mailman/listinfo/voiceops
>>>
>>
>>
>
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