IP Phone Does Not Initiate Registratio to SRST

[Boyan Sotirov]

I have been playing aroung with a pretty simple and straight forward setup. A CUCM, a phone which is registered there and an ISR 4351 router, which has to play the role of CUCM and a SRST device. 

The SIP dial peers are configured on the router, but when we try to simulate WAN failure, the phone does not even try to register on the SRST router (we verified that with a Wireshark trace from the phone port). 

I used the following link as a reference:

https://supportforums.cisco.com/t5/collaboration-voice-and-video/how-to-implement-cisco-unified-sip-srst/ta-p/3156033

According to the information provided in the guide, the phone must query the SRST router with a SIP REGISTER message to which the router shall reply with code 404. This does not happen in my case, the phone is completely ignorant of the SRST IP and does not send any requests while it's registered on the CUCM. 

I found this thread, where the problem is solved by fixing the SRST definition on the CUCM. But we did that already and we make sure that the we put the ip address of the SRST router for SIP.

https://supportforums.cisco.com/t5/ip-telephony/srst-sip/m-p/1866607#M180517 

On the IP phone, when we browse the menus in Settings-> Admin Settings->Security Setup->Trust List->Configuration - the IP address of the SRST router is listed there.

Also on the IP Phone Web Page we see the IP address of the SRST router listed as CM4 (CM1 and CM2 are the Publisher and Subscriber, CM3 is empty), there's no separate section for SRST on the Web Page. 

Is there anything we're missing?

voice service voip

address-hiding

mode border-element

allow-connections sip to sip

redirect ip2ip

sip

rel1xx disable

header-passing

error-passthru

registrar server expires max 600 min 60

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729r8

!

voice register global

default mode

no allow-hash-in-dn

system message "SIP SRST Mode"

max-dn 1000

max-pool 730

voice register pool 1

id network 192.168.5.0 mask 255.255.255.0

voice-class codec 1

no vad

[M02@rt37]

Hi @Boyan Sotirov

I can see on your voice service voip menu that the mode border element is activated. With CUBE activated, SIP SRST is not supported except if you have got on your ISR 4000 series the ios XE Fuji 16.7.1 Release. If you don't need CUBE functionalities then change your config or upgrade your ios.

See documention here to see the restriction (Feature Comparison on Supported Platforms)

https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/supported-platforms.html

[R0g22]

SIP SRST with CUBE co-located is supported starting 16.7.X Fuji XE release. Check the release notes for the new feature support. It is also stated in the same doc that you have linked.

[M02@rt37]

You're right @R0g22

He has got ISR 4000 series.

16.7.X Fuji XE release needed to co-located SIP SRST wuth CUBE.

[R0g22]

First thing, upgrade the IOS to 16.7.1 or above to support SIP SRST with CUBE co-located (if that is the case). If you are not using CUBE, there are a couple of configuration changes that I would suggest -

1. "voice service voip" does not have any SIP binds. Add it to the same IP that you have mentioned in the SRST reference on CUCM. This is mandatory only if the ISR has multiple interfaces configured with an IP address.
2. Lower the number of pools and DN's if you are not using that many phones/extensions.
3. Double check if your ID network config is good.

If you are still having issues with SIP SRST, enable the following and collect the logs during a SRST test -

debug ccsip message
debug ccsip non-call
debug ccsip error
debug voice register event
debug voice register error

[Boyan Sotirov]

Hello Nipun, 

The currently running IOS version is 16.7.2, that's the exact image name isr4300-universalk9.16.07.02.SPA.bin

And yes, we plan to use both SIP SRST and CUBE on the router. 

I have enabled the debugs, but again I don't see any SIP REGISTER message comig in from the phone. According to the documents I've read the Cisco IP phone should send SIP REGISTER message to the SRST router every couple of minutes, the SIP REGISTER message should have expire time 0, to which the SIP SRST router should respond with code 404. None of this happens. Even if there's something wrong with the SRST router, still what I cannot understand is why the phone does not even trying to reach the SIP SRST router.

And I bind all control and media communication to one of the router's interfaces, even though it's the only one that's active. But still, it does not work...

[R0g22]

The expires 0 REGISTER you talk about serves as a keep alive and only during non-srst mode.
In SRST, expires should be 3600 or whatever the expires is configured but not 0.
Can you take a pcap from the router interface and also take an output for -

show tcp brief all
show udp detail

[Boyan Sotirov · Accepted solution]

Hello Nipun, 

After stumbling around we found that we do not simulate properly the CUCM reachability failure. 

We were cutting off the uplink on the switch where the phone and the SRST router are connected. That's find, but the routing between the VLANs where the phone and the SRST router reside is done on the core switch :) So as happens in our case, the reachability to the CUCM is lost, the phone has its config for the SRST router, but it's in another network... but, the phone cannot reach it, because the ARP for its VLAN default gateway does not resolve, so it cannot send registration to the SRST router. And that's why we were not seeing any registration requests. We made a test where the phone and the SRST router are in the same L2 segment, hence no routing is needed - it works. So the Voice configs are in order, it was  just a routing/switching issue. To make a more valid test, we'll need to usa an ACL to block the traffic from the phone in question to the CUCM only. 

Thank you guys for being involved in this.

Regards,

Boyan

[R0g22]

Oh great you figured that out. :)

[Boyan Sotirov]

2. Lower the number of pools and DN's if you are not using that many phones/extensions. -> there's only one pool. Does it matter what resoursec are allocated when currently they're not being used and I don't see any signalling.
3. Double check if your ID network config is good. -> the network ID in the config is good, I verified it. I'm also capturing all in and out traffic from the port where the test phone is connected. That's the network.

[R0g22]

For 2, i was talking about the config under voice register global