11 Feb
2018
11 Feb
'18
11:37 p.m.
Hello,
I have found that calls are very inconsistent. I use Kamailio 5, Asterisk 14. When
certain Providers like (Sorenson, ZVRS) make calls into a WebRTC client (tryit-jssip),
sometime the calls stay up until I close them (10-15 minutes), others times those calls
drop in 30 seconds. This is extremely confusing...does anyone else experience this type
of behavior?
-Steve
12 Feb
12 Feb
2:27 a.m.
I have found that calls are very inconsistent. I use
Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of
moving parts with WebRTC. However, the typical reason why an established
call would drop after ~30 sec (32, to be precise) is that the end-to-end
ACK from the caller, which completes the required "three-way handshake"
for call establishment, does not reach the callee. This is because it's
not constructed correctly by the calling UA, not routed correctly by
intermediate entities, or isn't sent at all by the calling UA.
--
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/
3:06 a.m.
And just to add to Alex's comment -- in my experience this is almost always a firewall
issue. Whenever I see a call drop in 32 seconds, it is usually because a firewall did not
allow the ACK through.
Google "Asterisk drops calls after 32 seconds" and you get 1,180,000 results...
and the problem is not Asterisk itself.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, February 11, 2018 7:27 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
I have found that calls are very inconsistent. I use
Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of moving parts with
WebRTC. However, the typical reason why an established call would drop after ~30 sec (32,
to be precise) is that the end-to-end ACK from the caller, which completes the required
"three-way handshake"
for call establishment, does not reach the callee. This is because it's not
constructed correctly by the calling UA, not routed correctly by intermediate entities, or
isn't sent at all by the calling UA.
--
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/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:09 a.m.
This issue occurs when ACK is not routed to proper contact address.
On Sun, Feb 11, 2018 at 6:06 PM, Michael Young <myoung(a)redmonsters.net>
wrote:
And just to add to Alex's comment -- in my
experience this is almost
always a firewall issue. Whenever I see a call drop in 32 seconds, it is
usually because a firewall did not allow the ACK through.
Google "Asterisk drops calls after 32 seconds" and you get 1,180,000
results... and the problem is not Asterisk itself.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of
Alex Balashov
Sent: Sunday, February 11, 2018 7:27 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
--
Sincerely,
Brandon Armstead
CTO / CRYY.com
I have found that calls are very inconsistent. I
use Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of
moving parts with WebRTC. However, the typical reason why an established
call would drop after ~30 sec (32, to be precise) is that the end-to-end
ACK from the caller, which completes the required "three-way handshake"
for call establishment, does not reach the callee. This is because it's
not constructed correctly by the calling UA, not routed correctly by
intermediate entities, or isn't sent at all by the calling UA.
--
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/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
1:25 p.m.
But very often a call will stay up indefinitely! This is what is strange.
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Michael Young
Sent: Sunday, February 11, 2018 9:07 PM
To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
And just to add to Alex's comment -- in my experience this is almost always a firewall
issue. Whenever I see a call drop in 32 seconds, it is usually because a firewall did not
allow the ACK through.
Google "Asterisk drops calls after 32 seconds" and you get 1,180,000 results...
and the problem is not Asterisk itself.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, February 11, 2018 7:27 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
I have found that calls are very inconsistent. I use
Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of moving parts with
WebRTC. However, the typical reason why an established call would drop after ~30 sec (32,
to be precise) is that the end-to-end ACK from the caller, which completes the required
"three-way handshake"
for call establishment, does not reach the callee. This is because it's not
constructed correctly by the calling UA, not routed correctly by intermediate entities, or
isn't sent at all by the calling UA.
--
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/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
2:59 p.m.
Do that google search. You aren't the only person who experiences that.
For instance, it can be caused by having multiple UAs behind a NAT router\firewall with a
poor connection tracking implementation. The firewall can get confused and at times route
an ACK to the wrong UA (As Brandon Armstead said, "this issue occurs when ACK is not
routed to proper contact.") Without details on your topology, no one here would be
able to help you troubleshoot further or make other suggestions.
There is no substitute for packet capturing and a clear understanding of the call flow.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve
Sent: Monday, February 12, 2018 6:26 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
But very often a call will stay up indefinitely! This is what is strange.
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Michael Young
Sent: Sunday, February 11, 2018 9:07 PM
To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
And just to add to Alex's comment -- in my experience this is almost always a firewall
issue. Whenever I see a call drop in 32 seconds, it is usually because a firewall did not
allow the ACK through.
Google "Asterisk drops calls after 32 seconds" and you get 1,180,000 results...
and the problem is not Asterisk itself.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, February 11, 2018 7:27 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
I have found that calls are very inconsistent. I use
Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of moving parts with
WebRTC. However, the typical reason why an established call would drop after ~30 sec (32,
to be precise) is that the end-to-end ACK from the caller, which completes the required
"three-way handshake"
for call establishment, does not reach the callee. This is because it's not
constructed correctly by the calling UA, not routed correctly by intermediate entities, or
isn't sent at all by the calling UA.
--
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/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:11 p.m.
Thank you. Yes I have been doing packet capturing and it does appear that sometimes there
is a missing ACK back from Kamailio; it is just so, so confusing as to why sometimes
(often, depending the provider) it works. Your thoughts do help though!
-Steve
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Michael Young
Sent: Monday, February 12, 2018 8:59 AM
To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
Do that google search. You aren't the only person who experiences that.
For instance, it can be caused by having multiple UAs behind a NAT router\firewall with a
poor connection tracking implementation. The firewall can get confused and at times route
an ACK to the wrong UA (As Brandon Armstead said, "this issue occurs when ACK is not
routed to proper contact.") Without details on your topology, no one here would be
able to help you troubleshoot further or make other suggestions.
There is no substitute for packet capturing and a clear understanding of the call flow.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve
Sent: Monday, February 12, 2018 6:26 AM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
But very often a call will stay up indefinitely! This is what is strange.
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Michael Young
Sent: Sunday, February 11, 2018 9:07 PM
To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
And just to add to Alex's comment -- in my experience this is almost always a firewall
issue. Whenever I see a call drop in 32 seconds, it is usually because a firewall did not
allow the ACK through.
Google "Asterisk drops calls after 32 seconds" and you get 1,180,000 results...
and the problem is not Asterisk itself.
Michael
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, February 11, 2018 7:27 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
I have found that calls are very inconsistent. I use
Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of moving parts with
WebRTC. However, the typical reason why an established call would drop after ~30 sec (32,
to be precise) is that the end-to-end ACK from the caller, which completes the required
"three-way handshake"
for call establishment, does not reach the callee. This is because it's not
constructed correctly by the calling UA, not routed correctly by intermediate entities, or
isn't sent at all by the calling UA.
--
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/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
12:48 p.m.
But the weird part is that one call may stay up until I hang up(10 -15 minutes), the other
for 30 seconds, that's with using the same UA and no changes to any scripts.
-----Original Message-----
From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov
Sent: Sunday, February 11, 2018 8:27 PM
To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] Very inconsistent calls
I have found that calls are very inconsistent. I use
Kamailio 5,
Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls
into a WebRTC client (tryit-jssip), sometime the calls stay up until I
close them (10-15 minutes), others times those calls drop in 30
seconds. This is extremely confusing...does anyone else experience
this type of behavior?
It is hard to speculate without a capture, and indeed there are lots of moving parts with
WebRTC. However, the typical reason why an established call would drop after ~30 sec (32,
to be precise) is that the end-to-end ACK from the caller, which completes the required
"three-way handshake"
for call establishment, does not reach the callee. This is because it's not
constructed correctly by the calling UA, not routed correctly by intermediate entities, or
isn't sent at all by the calling UA.
--
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/
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
2279
days inactive
2280
days old
7 comments
4 participants
participants (4)
-
Alex Balashov -
Brandon Armstead -
Michael Young -
Wilkins, Steve