CIsco CUBE NO Audio Issue with SRTP RTP interworking

gary391 · ‎07-10-2024

Hi Team,

I am working to onboard CISCO cube device to our network. This is first cisco voice device I have ever worked on, i have very limited to no prior knowledge related to Cisco devices in general.

Setup

SBC <--TLS/SRTP--> CUBE <--> SIP trunk provider

Below is an outbound call from the SBC to CUBE to siptrunk (I have verified the media stream from the SBC side)

show call active voice brief

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
STCAPP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
243C : 3337 159916810ms.1 (*01:34:39.259 UTC Thu Jul 11 2024) +9490 pid:100 Answer +1425XXXXXXX active
dur 00:00:02 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.142.161.171:32154 SRTP: on rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00
LocalUUID:052fc55c6caa59c387f345dbc9fb4d42
RemoteUUID:750df40c2ca754309a056dc8349db2f5
VRF: NA
243C : 3338 159916880ms.1 (*01:34:39.329 UTC Thu Jul 11 2024) +9410 pid:200 Originate +1817XXXXXXX active
dur 00:00:02 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 168.86.144.204:19030 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00
LocalUUID:00000000000000000000000000000000
RemoteUUID:052fc55c6caa59c387f345dbc9fb4d42
VRF: NA

Below is an inbound call from the siptrunk to CUBE to SBC. I don't see any packet count on the cube.

Telephony call-legs: 0
SIP call-legs: 2
H323 call-legs: 0
Call agent controlled call-legs: 0
SCCP call-legs: 0
STCAPP call-legs: 0
Multicast call-legs: 0
Total call-legs: 2
2480 : 3390 162247150ms.1 (*02:13:29.607 UTC Thu Jul 11 2024) +17570 pid:201 Answer +1817XXXXXXX connected
dur 00:00:09 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 168.86.144.196:11946 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00
LocalUUID:1c211862c79f5de6be65a687e7fb99b1
RemoteUUID:4093a39d95555891899c6ca3a1d3e837
VRF: NA
2480 : 3391 162247160ms.1 (*02:13:29.617 UTC Thu Jul 11 2024) +17560 pid:101 Originate +1425XXXXXXX active
dur 00:00:09 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0xB8 video tos:0x0
IP 10.142.161.171:32160 SRTP: on rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw TextRelay: off Transcoded: No ICE: Off
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
long duration call detected:n long duration call duration:n/a timestamp:n/a
LostPacketRate:0.00 OutOfOrderRate:0.00
LocalUUID:4093a39d95555891899c6ca3a1d3e837
RemoteUUID:1c211862c79f5de6be65a687e7fb99b1
VRF: NA

---

dial-peer voice 200 voip
description +10digit to siptrunk
destination-pattern +1.T
session protocol sipv2
session target sip-server
voice-class codec 2
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte sip-kpml sip-notify
no vad
dial-peer voice 201 voip
description Incoming from siptrunk
session protocol sipv2
incoming called-number +1425XXXXXXX
voice-class codec 2
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
no vad
dial-peer voice 100 voip
description Incoming from SBC
session protocol sipv2
session transport tcp tls
incoming called-number +1.T
voice-class codec 2
voice-class sip srtp-crypto 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte sip-kpml sip-notify
srtp
no vad
dial-peer voice 101 voip
description Outgoing to SBC
destination-pattern ++1425XXXXXXX
session protocol sipv2
session target ipv4:10.142.161.XXX
session transport tcp tls
voice-class codec 2
voice-class sip srtp-crypto 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte sip-kpml sip-notify
srtp
no vad

---

address-hiding
mode border-element
allow-connections sip to sip
shutdown
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
trace
sip
header-passing
error-passthru
early-offer forced
no call service stop
audio forced
!
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
!
!
voice class srtp-crypto 1
crypto 1 AEAD_AES_256_GCM
crypto 2 AEAD_AES_128_GCM
crypto 3 AES_CM_128_HMAC_SHA1_80
crypto 4 AES_CM_128_HMAC_SHA1_32
!
!
!
!
!
!
!
voice-card 0/4
dsp services dspfarm
no watchdog
!
no license feature hseck9
license udi pid ISR4331/K9 sn FDO271905VP
license boot suite FoundationSuiteK9
license boot level uck9

Any pointer as to what might be causing this will be really helpful ?

Also, if remove the TLS & SRTP/RTP interworking on CUBE and disable TLS/SRTP on the SBC side. Both the above calls work.

b.winter · ‎07-10-2024

Is it only the packet count stats that bother you, or do you really have audio issues?
The title says, you have audio issues, but at the end, you write that everything works. So what's the problem here now?

Roger Kallberg · ‎07-11-2024

A few things, number one I’m a bit confused by your setup, why do you have an SBC facing your Cube? Cube is an SBC, so you’re having this setup SBC -> SBC -> ITSP. This seems quite odd to me, why would you need two SBCs back to back?

Number 2, it looks like you have turned off the voice service, at least the presence of the “shutdown” command under voice service voip seems to indicate that.

Number 3, change your dial peer configuration to use information in the VIA field to match the calls inbound instead of using incoming called-number. See this document for more information on that. Explain Cisco IOS and IOS XE Call Routing

Last thing, have you imported the certificates from the SBC and also passed your certificate to the SBC so that you can form the TLS connection?

gary391 · ‎07-11-2024

Thanks Roger for your response.

1. SBC is a cloud hosted solution. Cisco CUBE will also support PRI connections at the local office. For this testing we are sip trunk.

2. Corrected the "shutdown" command in "voice service voip"

3. Will change it make the change, thanks for suggestion.

4. Yes, I have imported the certificates from the SBC and also passed your certificate to the SBC. The TLS connection gets established successfully when the call is setup.

Is there a way to know details of the SRTP packets on Cisco CUBE?

Can this is related to transcoding on the CUBE ?

Roger Kallberg · ‎07-11-2024

I doubt that it is related to transcoding as the call works when you’re not using SRTP. I think that for anyone to be able to understand what’s going on you would have to share the output from these debugs running simultaneously, debug ccsip message, debug ccsip non-call and debug VoIP ccapi inout. Please collect the output and put it into a plain text file that you attach to your response so that we can have a look at what’s happening.