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Issues with SIP timers - Vonage ITSP

mclean.danny
Level 1
Level 1

‎05-12-2016 06:50 AM - edited ‎03-17-2019 06:53 AM

Hi community,

I am having issues with registration stability with Vonage as out ITSP. Our call flow is SIP Phone -> CM ->SIP trunk -> CUBE-> ITSP. We are registering to the ITSP successfully initially but are getting complaints of dropped calls and the ITSP is telling us registrations are dropping often. It doesn't appear to be an network issue because we have site to site tunnels that are active without issues. Vonage is asking me to lower the settings for min-se timer to below 45 seconds but the CUBE will not allow me to do so. They are adamant that they never have to change this to accomidate a CUBE and that they have a lot of trunks with CUBE's which I'm sure they do.

Does the CUBE provide an way for me to know when registrations drop? I have a TAC case open but the engineer seems useless. They keep pointing me to generic documentation that doesn't cover what I am looking for. Is there a syslog message that can be generated so I can send a trap to get notified? If not from the registrar can it be done from the dial peer level?

Thanks for any insight that can be provided.

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Deepak Mehta
VIP Alumni
VIP Alumni

‎05-12-2016 07:40 AM

Hi Danny ,

In RTMT you will get detailed SIP messages in real time session trace which might give you an idea  on the dropped calls.As per my knowlege the only way we can find regisratration cause is by looking at traces "debus ccsip messages" and "Debug ccsip all " during the failure scenerio.

Can you collect and attach here.thanks.

You can read this for monitoring options available for CUBE. http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/white_paper_c11-613550.html

You can monitor the SIP trunk status for the CUBE-

2.2.1       SIP Trunk Status

SIP trunk status is an important element of CUBE monitoring. SIP Trunk status can be monitored by configuring an out-of-dialog (OOD) SIP Options PING as a keepalive mechanism on the dial-peer(s) pointing towards the SIP Trunk, using the CLI example below.

dial-peer voice 100 voip

destination-pattern .T

voice-class sip options-keepalive up-interval 100 down-interval 50 retry 6

session protocol sipv2

session target ipv4:x.x.x.x

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