[cisco-voip] SIP Fail over

Thu Dec 20 18:05:38 EST 2018

Nate,

Good point.  Might be hard to finesse a fake stratum into the master
command, without accidentally messing with the selection process.  Stratum
number isn't the only criteria, and the process seems to be pretty complex.

Looks like we might not be able to have our cake and eat it too.

You must be referring to the following sentence from the CUCM SRND
<https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/netstruc.html#pgfId-1497943>
:

"Cisco highly recommends configuring the publisher to point to a Stratum-1,
Stratum-2, or Stratum-3 NTP server to ensure that the cluster time is
synchronized with an external time source."

On Thu, Dec 20, 2018 at 4:46 PM NateCCIE <nateccie at gmail.com> wrote:

> I think the lowest cucm will use is a 3?
>
> Sent from my iPhone
>
> On Dec 20, 2018, at 3:35 PM, Anthony Holloway <
> avholloway+cisco-voip at gmail.com> wrote:
>
> I have never seen that done before.  I like it!
>
> Just be careful hard coding your stratum to a value of 2 all the time.
> Instead it should be a relative value, higher than your reference clock.
> Or if you do want a one-size-fits-all stratum, 14 maybe
> <http://support.ntp.org/bin/view/Support/SelectingOffsiteNTPServers#Section_5.3.5.>
> ?
>
> Thanks for sharing that tip Ryan!
>
>
>
> On Thu, Dec 20, 2018 at 3:52 PM Ryan Huff <ryanhuff at outlook.com> wrote:
>
>> I like ntp master 2 as a fallback, to allow synchronization with the
>> local device clock in a DR/Outage scenario where I fail sync to the actual
>> reference clock
>>
>> Sent from my iPhone
>>
>> On Dec 20, 2018, at 14:51, Anthony Holloway <
>> avholloway+cisco-voip at gmail.com> wrote:
>>
>> It's very interesting to me the kinds of things people take for granted
>> and go a long time without ever being corrected, simply because the people
>> who know these things, think it's common knowledge.
>>
>> For example, I had a conversation with a senior collab person once, who
>> didn't know that region bit rate settings were a ceiling, and that a lower
>> bit rate could be negotiated.
>>
>> And as another example, Engineers who put *ntp master* on a router
>> because they think this makes the router an NTP server.
>>
>> And as one last example, Engineers who use the ^ symbol at the beginning
>> of a Dial Peer destination pattern, not knowing that destination patterns
>> are left justified implicitly.  Or alternatively, don't use the $ at the
>> end, effectively creating a "begins with" clause, when an "is exactly"
>> clause is desired.
>>
>> Someone should start a thread titled: What is something you found out
>> that you were wrong about for a long time?
>>
>> On Thu, Dec 20, 2018 at 1:14 PM Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
>>
>>>
>>> I’ll be honest. I didn’t know there was a difference.
>>>
>>> I’m guessing a SIP trunk to a third party app that is reported as down
>>> due to to sip option ping really is down and not some silly networking
>>> issue where an icmp ping was failing.
>>>
>>> This is good to know.
>>>
>>> And the last thing I will learn this year. ;)
>>>
>>>
>>>
>>> *-sent from mobile device-*
>>>
>>>
>>> *Lelio Fulgenzi, B.A.* | Senior Analyst
>>>
>>> Computing and Communications Services | University of Guelph
>>>
>>> Room 037 Animal Science & Nutrition Bldg | 50 Stone Rd E | Guelph, ON |
>>> N1G 2W1
>>>
>>> 519-824-4120 Ext. 56354 <519-824-4120;56354> | lelio at uoguelph.ca
>>>
>>>
>>>
>>> www.uoguelph.ca/ccs
>>> <https://nam03.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.uoguelph.ca%2Fccs&data=02%7C01%7C%7C10a7704d902d47a4bb2b08d666b49661%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C636809323128783040&sdata=GxQXEHhlPK1yANdvpNcSsoGGyv%2FCvHq5MUsKEzfp44w%3D&reserved=0> |
>>> @UofGCCS on Instagram, Twitter and Facebook
>>>
>>>
>>>
>>> [image: University of Guelph Cornerstone with Improve Life tagline]
>>>
>>> On Dec 20, 2018, at 1:01 PM, Anthony Holloway <
>>> avholloway+cisco-voip at gmail.com> wrote:
>>>
>>> Erik,
>>>
>>> That's an interesting insight.  It kind of sounds like you think ICMP
>>> Ping and SIP OPTIONS Ping are related, but they are completely different.
>>>
>>> Just because you cannot ICMP Ping the SIP Peer at L3, doesn't mean you
>>> cannot OPTIONs them.
>>>
>>> Am I understanding your thought process correctly?
>>>
>>> On Thu, Dec 20, 2018 at 11:53 AM Ryan Huff <ryanhuff at outlook.com> wrote:
>>>
>>>>
>>>>
>>>> Thanks,
>>>>
>>>> Ryan Huff, CCDP, CCNP
>>>> Cisco Certified Network and Design Professional
>>>>
>>>> ------------------------------
>>>> *From:* Ryan Huff <ryanhuff at outlook.com>
>>>> *Sent:* Thursday, December 20, 2018 12:46 PM
>>>> *To:* Erik Anderson
>>>> *Subject:* Re: [cisco-voip] SIP Fail over
>>>>
>>>> Not sure what kind of code you're working with but if its modern, you
>>>> could try server groups. Here is a snippet from one of mine (using AT&T
>>>> admitidly), sanitized for the NSA ...
>>>>
>>>>
>>>> *voice class server-group 100 *
>>>>
>>>> * ipv4 12.x.x.x preference 1 *
>>>>
>>>> * ipv4 12.x.x.x preference 2 *
>>>>
>>>> * ipv4 12.x.x.x preference 3 *
>>>>
>>>> * ipv4 12.x.x.x preference 1 *
>>>>
>>>> * description PSTN SIGNALING PEERS *
>>>>
>>>> *! *
>>>>
>>>> *voice class server-group 200 *
>>>>
>>>> * ipv4 10.x.x.x preference 3 *
>>>>
>>>> * ipv4 10.x.x.x preference 1 *
>>>>
>>>> * ipv4 10.x.x.x preference 2 *
>>>>
>>>> * description CUCM SIGNALING PEERS *
>>>>
>>>> *! *
>>>>
>>>> *voice class sip-options-keepalive 100 *
>>>>
>>>> * description PSTN HEARTBEAT *
>>>>
>>>> *! *
>>>>
>>>> *voice class sip-options-keepalive 200 *
>>>>
>>>> * description CCM HEARTBEAT *
>>>>
>>>> *! *
>>>> *{ .. other config .. }*
>>>>
>>>>
>>>> *dial-peer voice 100 voip *
>>>>
>>>> * description INGRESS/EGRESS WITH PSTN *
>>>>
>>>> * translation-profile outgoing PLUS1_STRIP *
>>>>
>>>> * huntstop *
>>>>
>>>> * destination-pattern A *
>>>>
>>>> * session protocol sipv2 *
>>>>
>>>> * session server-group 100 *
>>>>
>>>> * destination dpg 200 *
>>>>
>>>> * incoming uri via PSTN *
>>>>
>>>> * voice-class codec 1   *
>>>>
>>>> * voice-class sip options-ping 60 *
>>>>
>>>> * voice-class sip profiles 100 *
>>>>
>>>> * voice-class sip options-keepalive profile 100 *
>>>>
>>>> * voice-class sip bind control source-interface XXXX *
>>>>
>>>> * voice-class sip bind media source-interface XXXX *
>>>>
>>>> * dtmf-relay rtp-nte sip-notify *
>>>> * no vad*
>>>>
>>>> *! *
>>>>
>>>> *dial-peer voice 200 voip *
>>>>
>>>> * description INGRESS/EGRESS WITH CUCM *
>>>>
>>>> * translation-profile outgoing PLUS1_STRIP *
>>>>
>>>> * huntstop *
>>>>
>>>> * destination-pattern A *
>>>>
>>>> * session protocol sipv2 *
>>>>
>>>> * session server-group 200 *
>>>>
>>>> * destination dpg 100 *
>>>>
>>>> * incoming uri via CUCM *
>>>>
>>>> * voice-class codec 1   *
>>>>
>>>> * voice-class sip profiles 200 *
>>>>
>>>> * voice-class sip options-keepalive profile 200 *
>>>>
>>>> * voice-class sip bind control source-interface XXXX *
>>>>
>>>> * voice-class sip bind media source-interface XXXX *
>>>>
>>>> * dtmf-relay rtp-nte sip-notify *
>>>> * no vad*
>>>> *!*
>>>>
>>>> Thanks,
>>>>
>>>> Ryan Huff, CCDP, CCNP
>>>> Cisco Certified Network and Design Professional
>>>>
>>>> ------------------------------
>>>> *From:* Erik Anderson <erik.anderson.85 at gmail.com>
>>>> *Sent:* Thursday, December 20, 2018 12:37 PM
>>>> *To:* Ryan Huff
>>>> *Subject:* Re: [cisco-voip] SIP Fail over
>>>>
>>>> Ryan,
>>>>
>>>> Level 3 does not support options ping. If i try to ping the call
>>>> control IP it will always fail. There is a separate pingable address, but I
>>>> didnt think i could configure the options ping to use any address other
>>>> than the target.
>>>>
>>>> On Thu, Dec 20, 2018 at 11:34 AM Ryan Huff <ryanhuff at outlook.com>
>>>> wrote:
>>>>
>>>> Couldn't you just use voice class sip options/keepalives to mark when
>>>> the ITSP is down, so CUCM marks the trunk out of service and fails to the
>>>> next route group member immediately (ideally, your secondary CUBE)? Seems
>>>> like thats a more natural way of doing it versus using IP SLAs...
>>>>
>>>> Thanks,
>>>>
>>>> - Ryan
>>>> ------------------------------
>>>> *From:* cisco-voip <cisco-voip-bounces at puck.nether.net> on behalf of
>>>> Erik Anderson <erik.anderson.85 at gmail.com>
>>>> *Sent:* Thursday, December 20, 2018 12:03 PM
>>>> *To:* cisco-voip voyp list
>>>> *Subject:* [cisco-voip] SIP Fail over
>>>>
>>>> Morning Folks,
>>>>
>>>> We have implemented a new SIP solution with Level 3 and found that we
>>>> have outbound calling failover issues. When a CUBE loses its ability to
>>>> talk to its Level 3 Peer, but can still talk to CUCM outbound calls will
>>>> still connect to the CUBE, but fail connecting to Level 3. In turn CUCM
>>>> still thinks the call is connected since the CUCM SIP trunk remains up to
>>>> the CUBE.
>>>>
>>>>
>>>>
>>>> Architecture Notes:
>>>>
>>>>
>>>>
>>>> 4 Locations with 1 CUBE Each
>>>>
>>>> 4 CUCM SIP Trunks with each connecting to one of the 4 CUBEs
>>>>
>>>> 4 CUCM Route Groups with Various CUBE/SIP Trunks assigned a
>>>> Distribution Algorithm of Top Down
>>>>
>>>> Each CUBE has 2 SIP Peers
>>>>
>>>> Each CUBE can only talk to its respective SIP peer via its local Level
>>>> 3 Transport to reduce call control latency by not allowing it to use the
>>>> DMVPN backup network
>>>>
>>>> Level 3 does not support SIP Options Ping
>>>>
>>>> CUCM Trunks have SIP Options Ping enabled
>>>>
>>>>
>>>>
>>>> Call Flows:
>>>>
>>>>
>>>>
>>>> Working Flow:
>>>>
>>>>
>>>>
>>>> Phone ----> SLRG ----> Route Group Member #1 ----> CUBE SIP TRUNK ---->
>>>> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ----> Call
>>>> Completes
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> CUBE Failure:
>>>>
>>>>
>>>>
>>>> Phone ----> SLRG ---->
>>>>
>>>>          Route Group Member #1 ----> CUBE SIP TRUNK --X--> CUBE (CUCM
>>>> Cant Reach CUBE)
>>>>
>>>>
>>>>
>>>>         CUCM Routes Call to Next Route Group Member
>>>>
>>>>
>>>>
>>>>                               Route Group Member #2 ----> CUBE SIP
>>>> TRUNK ----> CUBE ----> Level 3 Transport ----> Level 3 SIP Peer #1/#2 ---->
>>>> Call Completes
>>>>
>>>>
>>>>
>>>> Level 3 Transport Failure/SIP Server Failure:
>>>>
>>>>
>>>>
>>>> Phone ----> SLRG ---->
>>>>
>>>>          Route Group Member #1 ----> CUBE SIP TRUNK ----> CUBE --X-->
>>>> Level 3 Transport (CUBE Cant Reach Level 3 SIP Server)
>>>>
>>>>
>>>>
>>>>         CUCM Thinks Call Connects since the CUBE accepts the call,
>>>> Phone gets dead air, never tries the next RG Member
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> My idea to fix this is to use an IPSLA to ping the pingable address on
>>>> the Level 3 SIP Servers. If both address are unreachable then shutdown the
>>>> CUCM Dial-Peers. This doesn’t sounds like the best way of fixing it, but it
>>>> should work.
>>>>
>>>>
>>>>
>>>> If any has any other better ideas please let me know.
>>>> --
>>>> Erik Anderson
>>>> Telecom Manager
>>>> Some Random Corp.
>>>>
>>>>
>>>>
>>>> --
>>>> Erik Anderson
>>>> _______________________________________________
>>>> cisco-voip mailing list
>>>> cisco-voip at puck.nether.net
>>>> https://puck.nether.net/mailman/listinfo/cisco-voip
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>>>>
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