TCP Signaling for SIP Signaling
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?
On 07/16/2017 10:28 PM, Colton Conor 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?
...
My (extremely small, internal use only) deployment uses TCP for signalling wherever possible. As you said, it seems to be more reliable and allows the use of TLS. I can't speak to the implications of a large scale deployment.
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?
It uses UDP, and that's intentional. The back-off and repeat characteristics of TCP, while useful for reliable transmissions of bulk data with no real-time requirements, are detrimental to real-time, low-latency communication media. A single lost packet would cause a huge hole in the media during back-off and retransmit only to deliver data that's old and has to be dropped by the consumer to restore the real-time synchronization with the sender. It's far better to just accept dropped media and either have a short hole in the media or attempt to conceal it somehow which many modern CODECs support. Some also support using forward error correction and data timespace scrambling to completely reconstruct the original data when there's a small amount of packet loss at the cost of slightly higher end-to-end latency. -- Brandon Martin
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? _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
The use of TCP on the access/customer side of the network is increasing because of: 1. Growing SIP message sizes and headers, and the UDP fragmentation it engenders; 2. WebRTC and concomitant/similar technologies and feature sets, which use TCP or TCP-encapsulated transports; 3. Growing requirements of TLS/crypto, especially at the "last mile". That said, TCP is certainly more resource-intensive, just by definition. There is a lot of connection state to be held by the OS's network stack. However, this isn't as relevant a consideration as it was in the days when the guidance about the resource trade-offs was written, due to the increase in RAM and processing power. And, as Peter E. said, there is the problem of replicating / holding onto TCP state through failover, which has not been satisfactorily solved at a reasonably universal level. UDP continues to hold sway not only because of custom, habit and inertia, but also because of its innate simplicity and resource economies at large scales of message processing. It's not necessarily an inappropriate business decision in 2017 to switch your customers 100% to TCP. It depends on what kind of business-level trade-offs you are willing to make with regard to infrastructure, resource consumption and availability. UDP will probably continue to obtain inside the service provider network core and in inter-carrier, strictly Class 4 interconnects/trunk interfaces. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
On Mon, Jul 17, 2017 at 12:06:59PM +0300, Nikolay Shopik wrote:
On 17/07/17 06:08, Alex Balashov wrote:
2. WebRTC and concomitant/similar technologies and feature sets, which use TCP or TCP-encapsulated transports;
I belive WebRTC use SCTPoverDTLSoverUDP
WebRTC does not mandate a specific signalling technology or transport. SIP over WebSocket (and therefore TCP) is common in SIP implementations. It is certainly not the only way to do whatever it is that people usually envision when they say "WebRTC", which is in just a collection of browser media APIs and very loosely-defined standards for the networking portion. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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?
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
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?
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
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?
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
If you have multiple SBC's (e.g. DNS SRV) then yeah, it would drop and need to re-register for both UDP and TCP. I was talking about primary/standby failover, which is more likely than a complete data center failure (**knocks on wood**). Are you referring to Polycoms for TCPpreferred? I think that's the default Polycom setting. On Mon, Jul 17, 2017 at 10:59 AM, Colton Conor <colton.conor at gmail.com> wrote:
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?
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
Pete, Still not quite understanding how multiple SBC's vs primary/standby fail-over is differnet? Wouldn't primary/standby fail-over require 2 SBCs? Yes, I am referring to Polycom's TCPpreferred setting. I just looked at Polycom Admin guide, the the default is DNSnaptr not TCPpreferred. However, in Broadsoft's recommended device config file it says use TCPpreferred. Also in Polycom's guide, they recommend Note: Use BLF with TCPpreferred transport Use this feature with TCPpreferred transport. I assume this is due to UDP framentation with large BLF packets. So sounds like both Polycom and Broadsoft agree TCPpreferred aka use TCP when possible is the best method, especially when using BLF. So the question is why do most NOT due this. I assume there has to be some reason not to. On Mon, Jul 17, 2017 at 10:06 AM, Pete Eisengrein <peeip989 at gmail.com> wrote:
If you have multiple SBC's (e.g. DNS SRV) then yeah, it would drop and need to re-register for both UDP and TCP. I was talking about primary/standby failover, which is more likely than a complete data center failure (**knocks on wood**). Are you referring to Polycoms for TCPpreferred? I think that's the default Polycom setting.
On Mon, Jul 17, 2017 at 10:59 AM, Colton Conor <colton.conor at gmail.com> wrote:
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?
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
So sounds like both Polycom and Broadsoft agree TCPpreferred aka use TCP when possible is the best method, especially when using BLF. So the question is why do most NOT due this. I assume there has to be some reason not to. I don't believe this is the case. TCP was recommended iff you were using BLF and had large packets. (Some of the SoundPoint line had an issue with UDP fragmentation, I don't believe the VVX line has this issue) I don't believe Polycom and BroadSoft are asserting that TCP is the best method overall. Mike Sent from my iPhone On Jul 17, 2017, at 9:42 AM, Colton Conor <colton.conor at gmail.com> wrote: Pete, Still not quite understanding how multiple SBC's vs primary/standby fail-over is differnet? Wouldn't primary/standby fail-over require 2 SBCs? Yes, I am referring to Polycom's TCPpreferred setting. I just looked at Polycom Admin guide, the the default is DNSnaptr not TCPpreferred. However, in Broadsoft's recommended device config file it says use TCPpreferred. Also in Polycom's guide, they recommend Note: Use BLF with TCPpreferred transport Use this feature with TCPpreferred transport. I assume this is due to UDP framentation with large BLF packets. So sounds like both Polycom and Broadsoft agree TCPpreferred aka use TCP when possible is the best method, especially when using BLF. So the question is why do most NOT due this. I assume there has to be some reason not to. On Mon, Jul 17, 2017 at 10:06 AM, Pete Eisengrein <peeip989 at gmail.com> wrote:
If you have multiple SBC's (e.g. DNS SRV) then yeah, it would drop and need to re-register for both UDP and TCP. I was talking about primary/standby failover, which is more likely than a complete data center failure (**knocks on wood**). Are you referring to Polycoms for TCPpreferred? I think that's the default Polycom setting.
On Mon, Jul 17, 2017 at 10:59 AM, Colton Conor <colton.conor at gmail.com> wrote:
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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UDP remains the "weapon of choice" for sip trunking applications and for many subscriber applications. However, feature-rich subscriber implementations frequently require SIP messages to be substantially larger than the (approximately) 1500 byte MTU (maximum transmission unit) supported in the public internet. UDP has a mechanism for dealing with messages greater than MTU size, but it doesn't work reliably for SIP over UDP in the public internet. When you need to go > 1500 bytes, you need to go to TCP. Although SIP registration state and SIP call state can be replicated over a pair of SBC's (Acme or Sonus), TCP state cannot be maintained. There is way to much going on at the driver level for that to be practical. When switchover occurs, the TCP "transmission control" parameters are stale. So, there will be a TCP RST, and the endpoints need to start over with the whole SYN / SYN,ACK "three-way handshake" drill. Although the subscriber device will in all likelihood need to restart the TCP session, the worst-case disruption will be only as long as the re-registration interval. And even then, only outbound calls (toward the subscriber device) will be affected. But even in that unlucky instance, voice mail should save the day. The Acme "TCP session leak" was fixed several years ago At least three. Any software having that leak has been End of Support for long time. If anybody would like further discussion, feel free to contact me directly. John S. Robinson jsr (at) communichanic dot com Communichanic Consultants, Inc. On 7/17/2017 08:32, Pete Eisengrein 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 <mailto: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 <mailto: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 <mailto: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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On Mon, Jul 17, 2017 at 02:30:41PM -0500, John S. Robinson wrote:
Although SIP registration state and SIP call state can be replicated over a pair of SBC's (Acme or Sonus), TCP state cannot be maintained. There is way to much going on at the driver level for that to be practical. When switchover occurs, the TCP "transmission control" parameters are stale. So, there will be a TCP RST, and the endpoints need to start over with the whole SYN / SYN,ACK "three-way handshake" drill.
I am not familiar intimately with the modalities of commercial SBC vendors' failover options these days, but it had been my understanding that Acme Packet can, in fact, replicate TCP session state at the OS network stack level and fail it over "relatively seamlessly". I am not sure offhand where I learned that or whether it is accurate. Regardless, it is certainly not a common denominator within the industry, and is a difficult problem to solve. Moreover, speaking from an open-source vantage point, I think there is a lot of concern that big, ambitious, capital-intensive investments in this type of failover may not pay off if the future holds that failover at that layer be done more commonly by "infrastructure", i.e. at the hypervisor level. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
John knows first hand that ACME is unable to do this seamlessly. He and I have spent countless hours working on issues due to Acme's inability to replicate TCP state. Ed Robbins On 7/17/17 7:09 PM, Alex Balashov wrote:
I am not familiar intimately with the modalities of commercial SBC vendors' failover options these days, but it had been my understanding that Acme Packet can, in fact, replicate TCP session state at the OS network stack level and fail it over "relatively seamlessly". I am not sure offhand where I learned that or whether it is accurate.
Regardless, it is certainly not a common denominator within the industry, and is a difficult problem to solve.
On Mon, Jul 17, 2017 at 10:22:20PM -0400, Ed Robbins wrote:
John knows first hand that ACME is unable to do this seamlessly. He and I have spent countless hours working on issues due to Acme's inability to replicate TCP state.
I see. Does the marketing literature claim they can do it? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
On 17/07/17 05:28, Colton Conor 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?
We switched to TCP 2 years ago since we need TLS. And experience with end-points was mostly positive. Not all of them quite same but its much improvement over UDP for us. pros - Better timeouts if you consider default UDP (where its 32s) - Server which reset your registration usualy send tcp.rst, which allow client to immediately start try re-register (not all of them do that) - protect you from async routing of packets (we had issue where we protect TCP stream with VPN tunnel, but at some point something broke at farend side and packets keep flowing around tunnel and UDP gladly accept them since there no session establishing) cons - Its still TCP and putting it on heavy loaded trunk will cause Head of Line blocking from time to time. Same applies to end-points if there is packet lost, but its negligible. RTP will still use UDP there is no need for TCP there it will make it just worse with its HoL problem.
On Mon, Jul 17, 2017 at 12:19:47PM +0300, Nikolay Shopik wrote:
- protect you from async routing of packets (we had issue where we protect TCP stream with VPN tunnel, but at some point something broke at farend side and packets keep flowing around tunnel and UDP gladly accept them since there no session establishing)
Assuming that firewall rules make this possible, wouldn't this occur with TCP as well, by way of the client or server simply opening a new TCP connection next time one needs to contact the other? -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
On 17/07/17 12:22, Alex Balashov wrote:
On Mon, Jul 17, 2017 at 12:19:47PM +0300, Nikolay Shopik wrote:
- protect you from async routing of packets (we had issue where we protect TCP stream with VPN tunnel, but at some point something broke at farend side and packets keep flowing around tunnel and UDP gladly accept them since there no session establishing)
Assuming that firewall rules make this possible, wouldn't this occur with TCP as well, by way of the client or server simply opening a new TCP connection next time one needs to contact the other?
Our case was with multi homing, TCP won't accept SYN.ACK response on different interface since its binding to IP when sending SYN. Unfortunately I don't remember all details now. So yeah TCP doesn't protect from all async cases but helps in some.
On 17/07/17 03:28, Colton Conor wrote:
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?
No harm keeping UDP. It is up to service providers what they want to use. And what works for their customers. I'm in favour of more TLS - it stops ALGs for sure.
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?
Normal SRTP is UDP. No reason not to run RTP over TCP. It's an option to configure in Snom phones. I think MS Lync might use it. If you get packet loss, retransmits would kill delay the call. Depends on packetloss vs round trip time. Tim
On 16 July 2017 at 19:28, Colton Conor <colton.conor at gmail.com> wrote:
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.
Knowing that TCP uses more overhead just by being TCP, is it really better for mobile phone batteries? -- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info
Are low re-registration intervals and flurries of OPTIONS pings and/or CRLF keepalives over UDP better for mobile batteries? :-) -- Alex
On Jul 17, 2017, at 5:12 PM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
On 16 July 2017 at 19:28, Colton Conor <colton.conor at gmail.com> wrote: 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.
Knowing that TCP uses more overhead just by being TCP, is it really better for mobile phone batteries?
-- ------- inum: 883510009027723 sip: jungleboogie at sip2sip.info _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
On 07/17/2017 05:12 PM, jungle Boogie wrote:
On 16 July 2017 at 19:28, Colton Conor <colton.conor at gmail.com> wrote:
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.
Knowing that TCP uses more overhead just by being TCP, is it really better for mobile phone batteries?
Anecdotal evidence using the built-in Android SIP client (which, while hilariously incapable, does support both UDP and TCP for signalling and is a readily-available common denominator) suggests that the less frequent need to refresh the registration and similar keepalive type mechanisms needed for UDP signalling does indeed make TCP noticeably better for mobile phone battery life.
Also, not knowing much about how mobile carrier data operates, I can't weigh in, but I suspect that the carrier-grade NAT and/or connection proxy implementations on the operator side have a great deal of influence on how well SIP on mobile over UDP works in practice. -- Alex -- Principal, Evariste Systems LLC (www.evaristesys.com) Sent from my Google Nexus.
Here is the theory behind "TCP for Mobile Phones" recommendations: * tcp - keep alive is supposed to happen from the Server side (SBC) therefore * application layer can go to sleep, and firewall pin hole is kept open at the driver level * thus conserving power needed to load and wake up the application. UAC's only need to register at the SIP application layer once every 10 minutes or so. In real life however, that doesn't seem to work out all that well due to SBC limitations (both Acme and Sonus; details on request) and limited support from the various phone app vendors. Plus, if the TCP connection "gets lost" the risk interval of not receiving a phone call is just that much longer. John On 7/17/2017 16:12, jungle Boogie wrote:
On 16 July 2017 at 19:28, Colton Conor <colton.conor at gmail.com> wrote:
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.
Knowing that TCP uses more overhead just by being TCP, is it really better for mobile phone batteries?
On 18/07/17 00:12, jungle Boogie wrote:
On 16 July 2017 at 19:28, Colton Conor <colton.conor at gmail.com> wrote:
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.
Knowing that TCP uses more overhead just by being TCP, is it really better for mobile phone batteries?
Some mobile clients (on both iOS and android) allow you to use push notification. Thus mobile client only keep TCP connection when app is open. And push server keep connection/registration, when app closed only to wake it up when need to accept call.
This is true. I still worry about the NAT issue when using UDP. Most all wireless carrier do not assign public facing IPv4 addresses to the phone only IPv6. On Tue, Jul 18, 2017 at 3:56 AM, Nikolay Shopik <shopik+lists at nvcube.net> wrote:
On 18/07/17 00:12, jungle Boogie wrote:
On 16 July 2017 at 19:28, Colton Conor <colton.conor at gmail.com> wrote:
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.
Knowing that TCP uses more overhead just by being TCP, is it really better for mobile phone batteries?
Some mobile clients (on both iOS and android) allow you to use push notification. Thus mobile client only keep TCP connection when app is open. And push server keep connection/registration, when app closed only to wake it up when need to accept call. _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
Le 2017-07-18 ? 09:51, Colton Conor a ?crit :
Most all wireless carrier do not assign public facing IPv4 addresses to the phone only IPv6.
What is the impact of this on UDP vs TCP? -- Simon Perreault Director of Engineering, Platform | Jive Communications, Inc. https://jive.com | +1 418 478 0989 ext. 1241 | sperreault at jive.com
I believe is you don't use SIP timers the session will close using UDP, whereas on TCP the session will stay active and keep the hole through the carriers firewall open to allow traffic to pass? On Tue, Jul 18, 2017 at 8:54 AM, Simon Perreault <sperreault at jive.com> wrote:
Le 2017-07-18 ? 09:51, Colton Conor a ?crit :
Most all wireless carrier do not assign public facing IPv4 addresses to the phone only IPv6.
What is the impact of this on UDP vs TCP?
-- Simon Perreault Director of Engineering, Platform | Jive Communications, Inc. https://jive.com | +1 418 478 0989 ext. 1241 | sperreault at jive.com
Le 2017-07-18 ? 09:58, Colton Conor a ?crit :
I believe is you don't use SIP timers the session will close using UDP, whereas on TCP the session will stay active and keep the hole through the carriers firewall open to allow traffic to pass?
NAT timeouts for TCP are generally much longer than for UDP, which allows keep-alives to be much less frequent, thus saving battery life. But that is true of any kind of NAT. Hence my question about NAT64's relevance to this discussion. -- Simon Perreault Director of Engineering, Platform | Jive Communications, Inc. https://jive.com | +1 418 478 0989 ext. 1241 | sperreault at jive.com
participants (11)
-
abalashov＠evaristesys.com -
colton.conor＠gmail.com -
ed＠erobbins.com -
jsr＠communichanic.com -
jungleboogie0＠gmail.com -
lists.voiceops＠monmotha.net -
mike＠redpoint.systems -
peeip989＠gmail.com -
shopik+lists＠nvcube.net -
sperreault＠jive.com -
tim＠kooky.org