overriding default timer values
Hello, The SIP RFC defines some timers and their default values. Is it possible to override these values in sip profiles or somewhere else? T1 500ms default Section 17.1.1.1 RTT Estimate T2 4s Section 17.1.2.2 The maximum retransmit interval for non-INVITE requests and INVITE responses T4 5s Section 17.1.2.2 Maximum duration a message will remain in the network Timer A initially T1 Section 17.1.1.2 INVITE request retransmit interval, for UDP only Timer B 64*T1 Section 17.1.1.2 INVITE transaction timeout timer Timer C > 3min Section 16.6 proxy INVITE transaction bullet 11 timeout Timer D > 32s for UDP Section 17.1.1.2 Wait time for response 0s for TCP/SCTP retransmits Timer E initially T1 Section 17.1.2.2 non-INVITE request retransmit interval, UDP only Timer F 64*T1 Section 17.1.2.2 non-INVITE transaction timeout timer Timer G initially T1 Section 17.2.1 INVITE response retransmit interval Timer H 64*T1 Section 17.2.1 Wait time for ACK receipt Timer I T4 for UDP Section 17.2.1 Wait time for 0s for TCP/SCTP ACK retransmits Timer J 64*T1 for UDP Section 17.2.2 Wait time for 0s for TCP/SCTP non-INVITE request retransmits Timer K T4 for UDP Section 17.1.2.2 Wait time for 0s for TCP/SCTP response retransmits
Yes. You can see these options in the stock SIP profile configuration XML: 1) <!-- Initial retransmission interval (in milliseconds). Set the T1 retransmission interval used by the SIP transaction engine. The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. --> <!-- <param name="timer-T1" value="500" /> --> 2) <!-- Transaction timeout (defaults to T1 * 64). Set the T1x64 timeout value used by the SIP transaction engine. The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine. The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. --> <!-- <param name="timer-T1X64" value="32000" /> --> 3) <!-- Maximum retransmission interval (in milliseconds). Set the maximum retransmission interval used by the SIP transaction engine. The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine. Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially until the timer B fires. --> <!-- <param name="timer-T2" value="4000" /> --> 4) <!-- Transaction lifetime (in milliseconds). Set the lifetime for completed transactions used by the SIP transaction engine. A completed transaction is kept around for the duration of T4 in order to catch late responses. The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. --> <!-- <param name="timer-T4" value="4000" /> --> -- Alex On 12/18/2013 04:37 AM, Grant Bagdasarian wrote:
Hello,
The SIP RFC defines some timers and their default values.
Is it possible to override these values in sip profiles or somewhere else?
T1 500ms default Section 17.1.1.1 RTT Estimate
T2 4s Section 17.1.2.2 The maximum retransmit
interval for non-INVITE
requests and INVITE
responses
T4 5s Section 17.1.2.2 Maximum duration a
message will
remain in the network
Timer A initially T1 Section 17.1.1.2 INVITE request retransmit
interval, for UDP only
Timer B 64*T1 Section 17.1.1.2 INVITE transaction
timeout timer
Timer C > 3min Section 16.6 proxy INVITE transaction
bullet 11 timeout
Timer D > 32s for UDP Section 17.1.1.2 Wait time for response
0s for TCP/SCTP retransmits
Timer E initially T1 Section 17.1.2.2 non-INVITE request
retransmit interval,
UDP only
Timer F 64*T1 Section 17.1.2.2 non-INVITE transaction
timeout timer
Timer G initially T1 Section 17.2.1 INVITE response
retransmit interval
Timer H 64*T1 Section 17.2.1 Wait time for
ACK receipt
Timer I T4 for UDP Section 17.2.1 Wait time for
0s for TCP/SCTP ACK retransmits
Timer J 64*T1 for UDP Section 17.2.2 Wait time for
0s for TCP/SCTP non-INVITE request
retransmits
Timer K T4 for UDP Section 17.1.2.2 Wait time for
0s for TCP/SCTP response retransmits
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Ahh, right! Thanks! Are these the only timers which can be overridden in FS? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Alex Balashov Sent: Wednesday, December 18, 2013 12:01 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] overriding default timer values Yes. You can see these options in the stock SIP profile configuration XML: 1) <!-- Initial retransmission interval (in milliseconds). Set the T1 retransmission interval used by the SIP transaction engine. The T1 is the initial duration used by request retransmission timers A and E (UDP) as well as response retransmission timer G. --> <!-- <param name="timer-T1" value="500" /> --> 2) <!-- Transaction timeout (defaults to T1 * 64). Set the T1x64 timeout value used by the SIP transaction engine. The T1x64 is duration used for timers B, F, H, and J (UDP) by the SIP transaction engine. The timeout value T1x64 can be adjusted separately from the initial retransmission interval T1. --> <!-- <param name="timer-T1X64" value="32000" /> --> 3) <!-- Maximum retransmission interval (in milliseconds). Set the maximum retransmission interval used by the SIP transaction engine. The T2 is the maximum duration used for the timers E (UDP) and G by the SIP transaction engine. Note that the timer A is not capped by T2. Retransmission interval of INVITE requests grows exponentially until the timer B fires. --> <!-- <param name="timer-T2" value="4000" /> --> 4) <!-- Transaction lifetime (in milliseconds). Set the lifetime for completed transactions used by the SIP transaction engine. A completed transaction is kept around for the duration of T4 in order to catch late responses. The T4 is the maximum duration for the messages to stay in the network and the duration of SIP timer K. --> <!-- <param name="timer-T4" value="4000" /> --> -- Alex
On 12/18/2013 06:40 AM, Grant Bagdasarian wrote:
Ahh, right!
Thanks!
Are these the only timers which can be overridden in FS?
As far as I can tell from looking at the configs and the source, yes. -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 United States Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/
Those settings cover everything but C and D. What else exactly are you trying to change and why? On Dec 18, 2013, at 6:45 AM, Alex Balashov <abalashov@evaristesys.com> wrote:
On 12/18/2013 06:40 AM, Grant Bagdasarian wrote:
Ahh, right!
Thanks!
Are these the only timers which can be overridden in FS?
As far as I can tell from looking at the configs and the source, yes.
I am running a multi tennant/Domain system using snom and grandstream phones. I use the BLF feature, works great except for one issue. If you have phones in two different domains with the same extension, the blfs show up accross the domains. So someone in domain 1 uses extension 20 or it rings, the blf's in domain 2 reflect the line in use. Any thoughts? Darcy Primrose Voice2net.
Make sure the user extension has a different sip_invite_domain and presence_id than the user extension in the other domain with something like: {sip_invite_domain=3.domain,presence_id=2000@3.domain }sofia/profile/blah@domain On Wed, Dec 18, 2013 at 1:19 PM, fs <fs@voice2net.ca> wrote:
I am running a multi tennant/Domain system using snom and grandstream phones. I use the BLF feature, works great except for one issue. If you have phones in two different domains with the same extension, the blfs show up accross the domains.
So someone in domain 1 uses extension 20 or it rings, the blf's in domain 2 reflect the line in use. Any thoughts?
Darcy Primrose Voice2net.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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Thanks to Moishe and Vik Great leads on the blf issue, I really appreciate the assistance. Darcy ----- Original Message ----- From: Vik Killa To: FreeSWITCH Users Help Sent: Wednesday, December 18, 2013 1:23 PM Subject: Re: [Freeswitch-users] BLF on Multi Domain Make sure the user extension has a different sip_invite_domain and presence_id than the user extension in the other domain with something like: {sip_invite_domain=3.domain,presence_id=2000@3.domain}sofia/profile/blah@domain On Wed, Dec 18, 2013 at 1:19 PM, fs <fs@voice2net.ca> wrote: I am running a multi tennant/Domain system using snom and grandstream phones. I use the BLF feature, works great except for one issue. If you have phones in two different domains with the same extension, the blfs show up accross the domains. So someone in domain 1 uses extension 20 or it rings, the blf's in domain 2 reflect the line in use. Any thoughts? Darcy Primrose Voice2net. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ No virus found in this message. Checked by AVG - www.avg.com Version: 2014.0.4259 / Virus Database: 3658/6930 - Release Date: 12/18/13
You are probably missing a context somewhere, I use blf with multi domain and never faced this issue. Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support@nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of fs Sent: Wednesday, December 18, 2013 1:19 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] BLF on Multi Domain I am running a multi tennant/Domain system using snom and grandstream phones. I use the BLF feature, works great except for one issue. If you have phones in two different domains with the same extension, the blfs show up accross the domains. So someone in domain 1 uses extension 20 or it rings, the blf's in domain 2 reflect the line in use. Any thoughts? Darcy Primrose Voice2net. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (6)
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Alex Balashov
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fs
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Grant Bagdasarian
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Michael Jerris
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Moishe Grunstein
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Vik Killa