Mailing List Archive: Cisco: VOIP

SIP Trunk Redundancy

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tednugent73 at gmail

Nov 23, 2009, 11:13 AM

Post #1 of 16 (1711 views)
Permalink

SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced by
SIP trunks. Everything appears to be running fine with the exception of
outbound trunk redundancy. The appear to have just removed the PRIs from the
existing RGs and replaced them with the SIP trunks. The problem is that if a
SIP trunk goes down its not rerouting to the next trunk, they are just
getting dead air. I'm assuming that this is similar to the issue seen with
H323 trunks and why a gatekeeper would be needed for this but what are the
options for SIP? I can probably get by with using Locations CAC for FO if
the trunks fills but not sure about if it actually goes down and CUCM can
determine that. CUCM 7.12 and no CUBE.

mh at markholloway

Nov 23, 2009, 11:18 AM

Post #2 of 16 (1680 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Dial Peer for SIP Trunk should use 'sip-server' (not DNS: or IP: statements) and the sip-ua would then be configured with the sip-server dns value.

sip-server support DNS SRV records. The ITSP should be providing multiple A records for DNS SRV query.

On Nov 23, 2009, at 12:13 PM, Ted Nugent wrote:

> I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

chrward at cisco

Nov 23, 2009, 11:21 AM

Post #3 of 16 (1676 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

You would need to look at the traces to verify, but it may just be the
time it takes to failover. You probably need to mess with the SIP
profiles and timers to get the trunks to failover in a timely manner. I
think by default it may take 15+ seconds (depends on # of retires and
time between retries) for a SIP trunk call to failover to the next
member of a route group.

-Chris

From: cisco-voip-bounces [at] puck
[mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced
by SIP trunks. Everything appears to be running fine with the exception
of outbound trunk redundancy. The appear to have just removed the PRIs
from the existing RGs and replaced them with the SIP trunks. The problem
is that if a SIP trunk goes down its not rerouting to the next trunk,
they are just getting dead air. I'm assuming that this is similar to the
issue seen with H323 trunks and why a gatekeeper would be needed for
this but what are the options for SIP? I can probably get by with using
Locations CAC for FO if the trunks fills but not sure about if it
actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.

lelio at uoguelph

Nov 23, 2009, 11:39 AM

Post #4 of 16 (1673 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Every day, I think to myself, man, SIP isn't all it's cracked up to be......

OK, not _every_ day, but when I read posts like this I do.

Seems like there will be some "re-edumacating" necessary when moving to SIP.

---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Bad grammar makes me [sic]" - Tshirt

----- Original Message -----
From: "Chris Ward (chrward)" <chrward [at] cisco>
To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <cisco-voip [at] puck>
Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
Subject: Re: [cisco-voip] SIP Trunk Redundancy

You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.

-Chris

From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.

_______________________________________________ cisco-voip mailing list cisco-voip [at] puck https://puck.nether.net/mailman/listinfo/cisco-voip

tednugent73 at gmail

Nov 23, 2009, 11:59 AM

Post #5 of 16 (1673 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Mark
Unfortunately there's not CUBE so no Dialpeers to make changes on, any other
ideas short of going with CUBE?

Chris
I'll try and pull some traces when we can test again, they have a funky
maintenance window so its hard to test anything except late at night. When
they first noticed this and got us engaged they said the sites primary SIP
routers power supply died and it never rolled to the next SIP Trunk, they
were able to physically reorder the routelist members to get outbound calls
working on the next RG in the RL but it appears to not to be rerouting on
its own? Is there an option I'm not seeing to enable trunk failover or
something like that?

Here are the current timers under the profile and what the defaults are set
to but if it didn't FO in well over an hour I'm thinking something else
might be at work here. Any thoughts as to which one specifically I'm looking
at so I can go in with a game plan?

Timer Invite Expires (seconds) = 180
Timer Register Delta (seconds) = 5
Timer Register Expires (seconds) = 3600
Timer T1 (msec) = 500
Timer T2 (msec) = 4000
Retry INVITE = 6
Retry Non-INVITE = 10

On Mon, Nov 23, 2009 at 2:21 PM, Chris Ward (chrward) <chrward [at] cisco>wrote:

> You would need to look at the traces to verify, but it may just be the
> time it takes to failover. You probably need to mess with the SIP profiles
> and timers to get the trunks to failover in a timely manner. I think by
> default it may take 15+ seconds (depends on # of retires and time between
> retries) for a SIP trunk call to failover to the next member of a route
> group.
>
>
>
> -Chris
>
>
>
> *From:* cisco-voip-bounces [at] puck [mailto:
> cisco-voip-bounces [at] puck] *On Behalf Of *Ted Nugent
> *Sent:* Monday, November 23, 2009 2:14 PM
> *To:* Cisco VoIPoE List
> *Subject:* [cisco-voip] SIP Trunk Redundancy
>
>
>
> I'm working with a client that has 3 sites where the PRIs were replaced by
> SIP trunks. Everything appears to be running fine with the exception of
> outbound trunk redundancy. The appear to have just removed the PRIs from the
> existing RGs and replaced them with the SIP trunks. The problem is that if a
> SIP trunk goes down its not rerouting to the next trunk, they are just
> getting dead air. I'm assuming that this is similar to the issue seen with
> H323 trunks and why a gatekeeper would be needed for this but what are the
> options for SIP? I can probably get by with using Locations CAC for FO if
> the trunks fills but not sure about if it actually goes down and CUCM can
> determine that. CUCM 7.12 and no CUBE.
>
>
>
>
>
>

tednugent73 at gmail

Nov 23, 2009, 12:01 PM

Post #6 of 16 (1677 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

I can't agree more Lelio...

On Mon, Nov 23, 2009 at 2:39 PM, Lelio Fulgenzi <lelio [at] uoguelph> wrote:

> Every day, I think to myself, man, SIP isn't all it's cracked up to
> be......
>
> OK, not _every_ day, but when I read posts like this I do.
>
> Seems like there will be some "re-edumacating" necessary when moving to
> SIP.
>
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Bad grammar makes me [sic]" - Tshirt
>
>
>
> ----- Original Message -----
> From: "Chris Ward (chrward)" <chrward [at] cisco>
> To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <
> cisco-voip [at] puck>
> Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [cisco-voip] SIP Trunk Redundancy
>
> You would need to look at the traces to verify, but it may just be the
> time it takes to failover. You probably need to mess with the SIP profiles
> and timers to get the trunks to failover in a timely manner. I think by
> default it may take 15+ seconds (depends on # of retires and time between
> retries) for a SIP trunk call to failover to the next member of a route
> group.
>
>
>
> -Chris
>
>
>
> *From:* cisco-voip-bounces [at] puck [mailto:
> cisco-voip-bounces [at] puck] *On Behalf Of *Ted Nugent
> *Sent:* Monday, November 23, 2009 2:14 PM
> *To:* Cisco VoIPoE List
> *Subject:* [cisco-voip] SIP Trunk Redundancy
>
>
>
> I'm working with a client that has 3 sites where the PRIs were replaced by
> SIP trunks. Everything appears to be running fine with the exception of
> outbound trunk redundancy. The appear to have just removed the PRIs from the
> existing RGs and replaced them with the SIP trunks. The problem is that if a
> SIP trunk goes down its not rerouting to the next trunk, they are just
> getting dead air. I'm assuming that this is similar to the issue seen with
> H323 trunks and why a gatekeeper would be needed for this but what are the
> options for SIP? I can probably get by with using Locations CAC for FO if
> the trunks fills but not sure about if it actually goes down and CUCM can
> determine that. CUCM 7.12 and no CUBE.
>
>
>
>
>
>
> _______________________________________________ cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

chrward at cisco

Nov 23, 2009, 12:03 PM

Post #7 of 16 (1674 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

I would set the retries for INVITES down to 1 or 2. Also, are you using
TCP or UDP?

-Chris

From: Ted Nugent [mailto:tednugent73 [at] gmail]
Sent: Monday, November 23, 2009 3:00 PM
To: Chris Ward (chrward)
Cc: Cisco VoIPoE List
Subject: Re: [cisco-voip] SIP Trunk Redundancy

Mark

Unfortunately there's not CUBE so no Dialpeers to make changes on, any
other ideas short of going with CUBE?

Chris

I'll try and pull some traces when we can test again, they have a funky
maintenance window so its hard to test anything except late at night.
When they first noticed this and got us engaged they said the sites
primary SIP routers power supply died and it never rolled to the next
SIP Trunk, they were able to physically reorder the routelist members to
get outbound calls working on the next RG in the RL but it appears to
not to be rerouting on its own? Is there an option I'm not seeing to
enable trunk failover or something like that?

Here are the current timers under the profile and what the defaults are
set to but if it didn't FO in well over an hour I'm thinking something
else might be at work here. Any thoughts as to which one specifically
I'm looking at so I can go in with a game plan?

Timer Invite Expires (seconds) = 180

Timer Register Delta (seconds) = 5

Timer Register Expires (seconds) = 3600

Timer T1 (msec) = 500

Timer T2 (msec) = 4000

Retry INVITE = 6

Retry Non-INVITE = 10

On Mon, Nov 23, 2009 at 2:21 PM, Chris Ward (chrward)
<chrward [at] cisco> wrote:

You would need to look at the traces to verify, but it may just be the
time it takes to failover. You probably need to mess with the SIP
profiles and timers to get the trunks to failover in a timely manner. I
think by default it may take 15+ seconds (depends on # of retires and
time between retries) for a SIP trunk call to failover to the next
member of a route group.

-Chris

From: cisco-voip-bounces [at] puck
[mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced
by SIP trunks. Everything appears to be running fine with the exception
of outbound trunk redundancy. The appear to have just removed the PRIs
from the existing RGs and replaced them with the SIP trunks. The problem
is that if a SIP trunk goes down its not rerouting to the next trunk,
they are just getting dead air. I'm assuming that this is similar to the
issue seen with H323 trunks and why a gatekeeper would be needed for
this but what are the options for SIP? I can probably get by with using
Locations CAC for FO if the trunks fills but not sure about if it
actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.

tednugent73 at gmail

Nov 23, 2009, 12:19 PM

Post #8 of 16 (1665 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

I will try changing the invites for sure.
I noticed that they are using a custom security profile and I just found out
that the provider is currently only accepting UDP for outgoing. Apparently
they had issues when they first set this up and the only way they could get
it working was to lock in the security profile with outgoing UDP. Think this
might be part of the problem?

On Mon, Nov 23, 2009 at 3:03 PM, Chris Ward (chrward) <chrward [at] cisco>wrote:

> I would set the retries for INVITES down to 1 or 2. Also, are you using
> TCP or UDP?
>
>
>
> -Chris
>
>
>
> *From:* Ted Nugent [mailto:tednugent73 [at] gmail]
> *Sent:* Monday, November 23, 2009 3:00 PM
> *To:* Chris Ward (chrward)
> *Cc:* Cisco VoIPoE List
> *Subject:* Re: [cisco-voip] SIP Trunk Redundancy
>
>
>
> Mark
>
> Unfortunately there's not CUBE so no Dialpeers to make changes on, any
> other ideas short of going with CUBE?
>
>
>
> Chris
>
> I'll try and pull some traces when we can test again, they have a funky
> maintenance window so its hard to test anything except late at night. When
> they first noticed this and got us engaged they said the sites primary SIP
> routers power supply died and it never rolled to the next SIP Trunk, they
> were able to physically reorder the routelist members to get outbound calls
> working on the next RG in the RL but it appears to not to be rerouting on
> its own? Is there an option I'm not seeing to enable trunk failover or
> something like that?
>
>
>
> Here are the current timers under the profile and what the defaults are set
> to but if it didn't FO in well over an hour I'm thinking something else
> might be at work here. Any thoughts as to which one specifically I'm looking
> at so I can go in with a game plan?
>
>
>
> Timer Invite Expires (seconds) = 180
>
> Timer Register Delta (seconds) = 5
>
> Timer Register Expires (seconds) = 3600
>
> Timer T1 (msec) = 500
>
> Timer T2 (msec) = 4000
>
> Retry INVITE = 6
>
> Retry Non-INVITE = 10
>
>
>
>
>
> On Mon, Nov 23, 2009 at 2:21 PM, Chris Ward (chrward) <chrward [at] cisco>
> wrote:
>
> You would need to look at the traces to verify, but it may just be the time
> it takes to failover. You probably need to mess with the SIP profiles and
> timers to get the trunks to failover in a timely manner. I think by default
> it may take 15+ seconds (depends on # of retires and time between retries)
> for a SIP trunk call to failover to the next member of a route group.
>
>
>
> -Chris
>
>
>
> *From:* cisco-voip-bounces [at] puck [mailto:
> cisco-voip-bounces [at] puck] *On Behalf Of *Ted Nugent
> *Sent:* Monday, November 23, 2009 2:14 PM
> *To:* Cisco VoIPoE List
> *Subject:* [cisco-voip] SIP Trunk Redundancy
>
>
>
> I'm working with a client that has 3 sites where the PRIs were replaced by
> SIP trunks. Everything appears to be running fine with the exception of
> outbound trunk redundancy. The appear to have just removed the PRIs from the
> existing RGs and replaced them with the SIP trunks. The problem is that if a
> SIP trunk goes down its not rerouting to the next trunk, they are just
> getting dead air. I'm assuming that this is similar to the issue seen with
> H323 trunks and why a gatekeeper would be needed for this but what are the
> options for SIP? I can probably get by with using Locations CAC for FO if
> the trunks fills but not sure about if it actually goes down and CUCM can
> determine that. CUCM 7.12 and no CUBE.
>
>
>
>
>
>
>

chrward at cisco

Nov 23, 2009, 12:29 PM

Post #9 of 16 (1666 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

If it were TCP it might be fast as the TCP connection would time out
much faster. For UDP, let's start with just reducing the number of
retries. The SIP Profile may not do this as I recall, the SIP profile is
used primarily for SIP end points. There are SIP CCM Service parameters
that should cover this. Perhaps in the service window you can try both.

-Chris

From: Ted Nugent [mailto:tednugent73 [at] gmail]
Sent: Monday, November 23, 2009 3:20 PM
To: Chris Ward (chrward)
Cc: Cisco VoIPoE List
Subject: Re: [cisco-voip] SIP Trunk Redundancy

I will try changing the invites for sure.

I noticed that they are using a custom security profile and I just found
out that the provider is currently only accepting UDP for outgoing.
Apparently they had issues when they first set this up and the only way
they could get it working was to lock in the security profile with
outgoing UDP. Think this might be part of the problem?

On Mon, Nov 23, 2009 at 3:03 PM, Chris Ward (chrward)
<chrward [at] cisco> wrote:

I would set the retries for INVITES down to 1 or 2. Also, are you using
TCP or UDP?

-Chris

From: Ted Nugent [mailto:tednugent73 [at] gmail]
Sent: Monday, November 23, 2009 3:00 PM
To: Chris Ward (chrward)
Cc: Cisco VoIPoE List
Subject: Re: [cisco-voip] SIP Trunk Redundancy

Mark

Unfortunately there's not CUBE so no Dialpeers to make changes on, any
other ideas short of going with CUBE?

Chris

I'll try and pull some traces when we can test again, they have a funky
maintenance window so its hard to test anything except late at night.
When they first noticed this and got us engaged they said the sites
primary SIP routers power supply died and it never rolled to the next
SIP Trunk, they were able to physically reorder the routelist members to
get outbound calls working on the next RG in the RL but it appears to
not to be rerouting on its own? Is there an option I'm not seeing to
enable trunk failover or something like that?

Here are the current timers under the profile and what the defaults are
set to but if it didn't FO in well over an hour I'm thinking something
else might be at work here. Any thoughts as to which one specifically
I'm looking at so I can go in with a game plan?

Timer Invite Expires (seconds) = 180

Timer Register Delta (seconds) = 5

Timer Register Expires (seconds) = 3600

Timer T1 (msec) = 500

Timer T2 (msec) = 4000

Retry INVITE = 6

Retry Non-INVITE = 10

On Mon, Nov 23, 2009 at 2:21 PM, Chris Ward (chrward)
<chrward [at] cisco> wrote:

You would need to look at the traces to verify, but it may just be the
time it takes to failover. You probably need to mess with the SIP
profiles and timers to get the trunks to failover in a timely manner. I
think by default it may take 15+ seconds (depends on # of retires and
time between retries) for a SIP trunk call to failover to the next
member of a route group.

-Chris

From: cisco-voip-bounces [at] puck
[mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced
by SIP trunks. Everything appears to be running fine with the exception
of outbound trunk redundancy. The appear to have just removed the PRIs
from the existing RGs and replaced them with the SIP trunks. The problem
is that if a SIP trunk goes down its not rerouting to the next trunk,
they are just getting dead air. I'm assuming that this is similar to the
issue seen with H323 trunks and why a gatekeeper would be needed for
this but what are the options for SIP? I can probably get by with using
Locations CAC for FO if the trunks fills but not sure about if it
actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.

mh at markholloway

Nov 23, 2009, 4:20 PM

Post #10 of 16 (1648 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

SIP Trunking works great when designed and deployed correctly.

On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:

> Every day, I think to myself, man, SIP isn't all it's cracked up to be......
>
> OK, not _every_ day, but when I read posts like this I do.
>
> Seems like there will be some "re-edumacating" necessary when moving to SIP.
>
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Bad grammar makes me [sic]" - Tshirt
>
>
> ----- Original Message -----
> From: "Chris Ward (chrward)" <chrward [at] cisco>
> To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <cisco-voip [at] puck>
> Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [cisco-voip] SIP Trunk Redundancy
>
> You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.
>
> -Chris
>
> From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
> Sent: Monday, November 23, 2009 2:14 PM
> To: Cisco VoIPoE List
> Subject: [cisco-voip] SIP Trunk Redundancy
>
> I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.
>
>
>
>
> _______________________________________________ cisco-voip mailing list cisco-voip [at] puck https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip

lelio at uoguelph

Nov 23, 2009, 4:40 PM

Post #11 of 16 (1648 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

I'm sure it does...that's what I meant by the re-education part. It doesn't look like it will be a simple drop-in replacement. What ever is, eh?

---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Bad grammar makes me [sic]" - Tshirt

----- Original Message -----
From: "Mark Holloway" <mh [at] markholloway>
To: "Lelio Fulgenzi" <lelio [at] uoguelph>
Cc: "Chris Ward (chrward)" <chrward [at] cisco>, "Cisco VoIPoE List" <cisco-voip [at] puck>
Sent: Monday, November 23, 2009 7:20:51 PM GMT -05:00 US/Canada Eastern
Subject: Re: [cisco-voip] SIP Trunk Redundancy

SIP Trunking works great when designed and deployed correctly.

On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:

Every day, I think to myself, man, SIP isn't all it's cracked up to be......

OK, not _every_ day, but when I read posts like this I do.

Seems like there will be some "re-edumacating" necessary when moving to SIP.

---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Bad grammar makes me [sic]" - Tshirt

----- Original Message -----
From: "Chris Ward (chrward)" < chrward [at] cisco >
To: "Ted Nugent" < tednugent73 [at] gmail >, "Cisco VoIPoE List" < cisco-voip [at] puck >
Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
Subject: Re: [cisco-voip] SIP Trunk Redundancy

You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.

-Chris

From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.

_______________________________________________ cisco-voip mailing list cisco-voip [at] puck https://puck.nether.net/mailman/listinfo/cisco-voip _______________________________________________
cisco-voip mailing list
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thomaslemay at comcast

Nov 23, 2009, 4:49 PM

Post #12 of 16 (1654 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Hi, Mark,

Can you suggest any best business practices for the design and deployment of
SIP trunking? Can you point me to any good books on SIP design and
deployment?

Thanks,

Tom

_____

From: cisco-voip-bounces [at] puck
[mailto:cisco-voip-bounces [at] puck] On Behalf Of Mark Holloway
Sent: Monday, November 23, 2009 7:21 PM
To: Lelio Fulgenzi
Cc: Cisco VoIPoE List
Subject: Re: [cisco-voip] SIP Trunk Redundancy

SIP Trunking works great when designed and deployed correctly.

On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:

Every day, I think to myself, man, SIP isn't all it's cracked up to be......

OK, not _every_ day, but when I read posts like this I do.

Seems like there will be some "re-edumacating" necessary when moving to SIP.

---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Bad grammar makes me [sic]" - Tshirt

----- Original Message -----
From: "Chris Ward (chrward)" <chrward [at] cisco>
To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List"
<cisco-voip [at] puck>
Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
Subject: Re: [cisco-voip] SIP Trunk Redundancy

You would need to look at the traces to verify, but it may just be the time
it takes to failover. You probably need to mess with the SIP profiles and
timers to get the trunks to failover in a timely manner. I think by default
it may take 15+ seconds (depends on # of retires and time between retries)
for a SIP trunk call to failover to the next member of a route group.

-Chris

From: cisco-voip-bounces [at] puck
[mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced by
SIP trunks. Everything appears to be running fine with the exception of
outbound trunk redundancy. The appear to have just removed the PRIs from the
existing RGs and replaced them with the SIP trunks. The problem is that if a
SIP trunk goes down its not rerouting to the next trunk, they are just
getting dead air. I'm assuming that this is similar to the issue seen with
H323 trunks and why a gatekeeper would be needed for this but what are the
options for SIP? I can probably get by with using Locations CAC for FO if
the trunks fills but not sure about if it actually goes down and CUCM can
determine that. CUCM 7.12 and no CUBE.

_______________________________________________ cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

tednugent73 at gmail

Nov 23, 2009, 4:50 PM

Post #13 of 16 (1647 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Unfortunately with providers selling SIP trunks and
informing their customers that CUBE is an expensive and unnecessary
accessory that Cisco uses to sell bigger and badder boxes it makes the
"design phase" difficult to say the least. Unless you live in a vacuum you
have to roll with the punches and use duct tape and bubble gum when you need
to or work for an organization that has a sales philosophy that allows you
to walk away from problematic and inflexible opportunities... I'm working on
the later.

On Mon, Nov 23, 2009 at 7:20 PM, Mark Holloway <mh [at] markholloway> wrote:

> SIP Trunking works great when designed and deployed correctly.
>
> On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:
>
> Every day, I think to myself, man, SIP isn't all it's cracked up to
> be......
>
> OK, not _every_ day, but when I read posts like this I do.
>
> Seems like there will be some "re-edumacating" necessary when moving to
> SIP.
>
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Bad grammar makes me [sic]" - Tshirt
>
>
> ----- Original Message -----
> From: "Chris Ward (chrward)" <chrward [at] cisco>
> To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <
> cisco-voip [at] puck>
> Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [cisco-voip] SIP Trunk Redundancy
>
> You would need to look at the traces to verify, but it may just be the time
> it takes to failover. You probably need to mess with the SIP profiles and
> timers to get the trunks to failover in a timely manner. I think by default
> it may take 15+ seconds (depends on # of retires and time between retries)
> for a SIP trunk call to failover to the next member of a route group.
>
>
> -Chris
>
>
> *From:* cisco-voip-bounces [at] puck [mailto:
> cisco-voip-bounces [at] puck] *On Behalf Of *Ted Nugent
> *Sent:* Monday, November 23, 2009 2:14 PM
> *To:* Cisco VoIPoE List
> *Subject:* [cisco-voip] SIP Trunk Redundancy
>
>
> I'm working with a client that has 3 sites where the PRIs were replaced by
> SIP trunks. Everything appears to be running fine with the exception of
> outbound trunk redundancy. The appear to have just removed the PRIs from the
> existing RGs and replaced them with the SIP trunks. The problem is that if a
> SIP trunk goes down its not rerouting to the next trunk, they are just
> getting dead air. I'm assuming that this is similar to the issue seen with
> H323 trunks and why a gatekeeper would be needed for this but what are the
> options for SIP? I can probably get by with using Locations CAC for FO if
> the trunks fills but not sure about if it actually goes down and CUCM can
> determine that. CUCM 7.12 and no CUBE.
>
>
>
>
>
> _______________________________________________ cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
>
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>
>

mh at markholloway

Nov 23, 2009, 5:58 PM

Post #14 of 16 (1656 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Cisco Press has a SIP Trunking book coming out in January 2010.

It's really up to the ITSP to test and certify each IP PBX for their SIP network. It's a painful process but if done correctly, it will work as expected and customers are happy.

On Nov 23, 2009, at 5:49 PM, Thomas LeMay wrote:

> Hi, Mark,
>
> Can you suggest any best business practices for the design and deployment of SIP trunking? Can you point me to any good books on SIP design and deployment?
>
> Thanks,
>
> Tom
>
> From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Mark Holloway
> Sent: Monday, November 23, 2009 7:21 PM
> To: Lelio Fulgenzi
> Cc: Cisco VoIPoE List
> Subject: Re: [cisco-voip] SIP Trunk Redundancy
>
> SIP Trunking works great when designed and deployed correctly.
>
> On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:
>
>
> Every day, I think to myself, man, SIP isn't all it's cracked up to be......
>
> OK, not _every_ day, but when I read posts like this I do.
>
> Seems like there will be some "re-edumacating" necessary when moving to SIP.
>
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> "Bad grammar makes me [sic]" - Tshirt
>
>
> ----- Original Message -----
> From: "Chris Ward (chrward)" <chrward [at] cisco>
> To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <cisco-voip [at] puck>
> Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [cisco-voip] SIP Trunk Redundancy
>
> You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.
>
> -Chris
>
> From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
> Sent: Monday, November 23, 2009 2:14 PM
> To: Cisco VoIPoE List
> Subject: [cisco-voip] SIP Trunk Redundancy
>
> I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.
>
>
>
>
> _______________________________________________ cisco-voip mailing list cisco-voip [at] puck https://puck.nether.net/mailman/listinfo/cisco-voip
> _______________________________________________
> cisco-voip mailing list
> cisco-voip [at] puck
> https://puck.nether.net/mailman/listinfo/cisco-voip
>

SCASPER at mtb

Nov 24, 2009, 4:24 AM

Post #15 of 16 (1628 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

We are testing a centralized SIP trunk design with CUBE and so far it seems to work well for basic inbound and outbound calling. We have over 1000 locations that range in size from 4 analog lines to multiple PRIs.Couple of questions about your SIP deployments:

Are you using SIP Trunk to support Fax? There does not seems to be a good way to ensure G711 is used for both inbound and outbound fax calls.

How are you handling 911 calls? With a centralized design I think CER would be required.since users have been known to pickup their phones and relocate without letting anyone know.

Of course both 911 and FAX could be served using separate analog lines or a PRI but that kind of defeats the purpose.

Steve

>>> Mark Holloway <mh [at] markholloway> 11/23/2009 7:20 PM >>>
SIP Trunking works great when designed and deployed correctly.

On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:

Every day, I think to myself, man, SIP isn't all it's cracked up to be......

OK, not _every_ day, but when I read posts like this I do.

Seems like there will be some "re-edumacating" necessary when moving to SIP.

---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
"Bad grammar makes me [sic]" - Tshirt

----- Original Message -----
From: "Chris Ward (chrward)" <chrward [at] cisco>
To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <cisco-voip [at] puck>
Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
Subject: Re: [cisco-voip] SIP Trunk Redundancy

You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.

-Chris

From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
Sent: Monday, November 23, 2009 2:14 PM
To: Cisco VoIPoE List
Subject: [cisco-voip] SIP Trunk Redundancy

I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.

_______________________________________________ cisco-voip mailing list cisco-voip [at] puck https://puck.nether.net/mailman/listinfo/cisco-voip
_______________________________________________
cisco-voip mailing list
cisco-voip [at] puck
https://puck.nether.net/mailman/listinfo/cisco-voip

************************************
This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
************************************

mh at markholloway

Nov 24, 2009, 8:46 AM

Post #16 of 16 (1621 views)
Permalink

Re: SIP Trunk Redundancy [In reply to]

Most SIP carriers support T.38 and/or G.711. The reliability of G.711 fax is mostly dependent upon the ITSP's media gateway platform.

The company I work for is using Broadworks as the SIP softswitch platform. The translations are built in such a way that calls will route to the appropriate PSAP no matter what SIP trunk they ingress. We look up based on the originating number and determine which PSAP to route to. The only time we have a concern are the customers who want out of LATA numbers. For example, someone physically resides in San Diego but wants L.A. phone numbers natively on their PBX. If they dial 911 the call will route to an L.A. PSAP. Broadworks has ways to accommodate this by assigning a separate number to the SIP trunk when anyone dials an emergency number. In this case, we would assign a native San Diego DN to the SIP trunk and when anyone utilizing the SIP trunk dials 911 the "From" number will always be the local San Diego number and route to the appropriate PSAP.

Some SIP carriers also offer 911 services through Intrado.

On Nov 24, 2009, at 5:24 AM, STEVEN CASPER wrote:

> We are testing a centralized SIP trunk design with CUBE and so far it seems to work well for basic inbound and outbound calling. We have over 1000 locations that range in size from 4 analog lines to multiple PRIs.Couple of questions about your SIP deployments:
>
> Are you using SIP Trunk to support Fax? There does not seems to be a good way to ensure G711 is used for both inbound and outbound fax calls.
>
> How are you handling 911 calls? With a centralized design I think CER would be required.since users have been known to pickup their phones and relocate without letting anyone know.
>
> Of course both 911 and FAX could be served using separate analog lines or a PRI but that kind of defeats the purpose.
>
> Steve
>
>
> SIP Trunking works great when designed and deployed correctly.
>
> On Nov 23, 2009, at 12:39 PM, Lelio Fulgenzi wrote:
>
>> Every day, I think to myself, man, SIP isn't all it's cracked up to be......
>>
>> OK, not _every_ day, but when I read posts like this I do.
>>
>> Seems like there will be some "re-edumacating" necessary when moving to SIP.
>>
>>
>> ---
>> Lelio Fulgenzi, B.A.
>> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
>> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
>> "Bad grammar makes me [sic]" - Tshirt
>>
>>
>> ----- Original Message -----
>> From: "Chris Ward (chrward)" <chrward [at] cisco>
>> To: "Ted Nugent" <tednugent73 [at] gmail>, "Cisco VoIPoE List" <cisco-voip [at] puck>
>> Sent: Monday, November 23, 2009 2:21:05 PM GMT -05:00 US/Canada Eastern
>> Subject: Re: [cisco-voip] SIP Trunk Redundancy
>>
>> You would need to look at the traces to verify, but it may just be the time it takes to failover. You probably need to mess with the SIP profiles and timers to get the trunks to failover in a timely manner. I think by default it may take 15+ seconds (depends on # of retires and time between retries) for a SIP trunk call to failover to the next member of a route group.
>>
>> -Chris
>>
>> From: cisco-voip-bounces [at] puck [mailto:cisco-voip-bounces [at] puck] On Behalf Of Ted Nugent
>> Sent: Monday, November 23, 2009 2:14 PM
>> To: Cisco VoIPoE List
>> Subject: [cisco-voip] SIP Trunk Redundancy
>>
>> I'm working with a client that has 3 sites where the PRIs were replaced by SIP trunks. Everything appears to be running fine with the exception of outbound trunk redundancy. The appear to have just removed the PRIs from the existing RGs and replaced them with the SIP trunks. The problem is that if a SIP trunk goes down its not rerouting to the next trunk, they are just getting dead air. I'm assuming that this is similar to the issue seen with H323 trunks and why a gatekeeper would be needed for this but what are the options for SIP? I can probably get by with using Locations CAC for FO if the trunks fills but not sure about if it actually goes down and CUCM can determine that. CUCM 7.12 and no CUBE.
>>
>>
>>
>>
>> _______________________________________________ cisco-voip mailing list cisco-voip [at] puck https://puck.nether.net/mailman/listinfo/cisco-voip
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip [at] puck
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> ************************************
> This email may contain privileged and/or confidential information that is intended solely for the use of the addressee. If you are not the intended recipient or entity, you are strictly prohibited from disclosing, copying, distributing or using any of the information contained in the transmission. If you received this communication in error, please contact the sender immediately and destroy the material in its entirety, whether electronic or hard copy. This communication may contain nonpublic personal information about consumers subject to the restrictions of the Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not directly or indirectly reuse or disclose such information for any purpose other than to provide the services for which you are receiving the information.
> There are risks associated with the use of electronic transmission. The sender of this information does not control the method of transmittal or service providers and assumes no duty or obligation for the security, receipt, or third party interception of this transmission.
> ************************************

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