Cisco IOS Cube One-Way-Audio

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G-Wire

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Hallo,

ich virtualisiere momentan Cisco IOS XE Software, Version 16.07.01
Auf dem Host, auf dem QEMU läuft, ist eine VPN Verbindung aktiv. Wireguard (Layer 3)
Das Problem: One-Way-Audio! Unten ist meine Topologie aufgeführt. An welcher Stelle ist das Problem zu beheben? Ich denke das Problem liegt vollständig auf der Seite des Cubes.


VM Host:
192.168.178.0/24 > Wireguard 10.0.0.0/24

Cisco IOS Cube:
GigabitEthernet1: 10.0.2.15/32
GigabitEthernet2: 192.168.20.1/32

VPN Server:
10.0.0.1/32

sip.conf:
Code: 
[general]
context=outgoing
bindport=5060
bindaddr=0.0.0.0
tcpenable=yes
transport=tcp
srvlookup=yes
dtmfmode=inband
externip=ÖFFENTLICHIP (aus Datenschutzgründen entfernt)
nat=force_rport,comedia
localnet=192.168.64.0/255.255.255.0
localnet=192.168.20.0/255.255.255.0
localnet=192.168.178.0/255.255.255.0
localnet=10.0.0.0/255.255.255.0
localnet=10.0.2.0/255.255.255.0

[9010]
type=friend
transport=udp
host=10.0.0.3
context=from-cube
allow=alaw
qualify=yes
;sendrpid=rpid
trustrpid=yes
dtmfmode=rfc2833
canreinvite=no
directmedia=no
secret=cisco123
fromuser=9010
defaultuser=9010
realm=asterisk
Der Cube kann nur eine Verbindung zu Asterisk herstellen, wenn host=10.0.0.3 gesetzt ist.
Bei host=dynamic oder einer anderen lokalen IP des Cubes kommt keine Registrierung zustande.
Im Kontext General habe ich unter Localnet sämtliche lokale IP Adressen eingetragen, da ich mir an dieser Stelle unsicher bin.

wg0.conf (WG Interface):
Code: 
...
[Peer]
PublicKey = XXX
AllowedIPs = 10.0.0.3/32, 192.168.20.1/32, 10.0.2.15/32

Cube:
Code: 
Router#show ip interface brief
Interface              IP-Address      OK? Method Status                Protocol
GigabitEthernet1       10.0.2.15       YES TFTP   up                    up
GigabitEthernet2       192.168.20.1    YES TFTP   up                    up

Anruf:
Code: 
*Dec  6 00:13:46.223: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6c9e3548;rport
Max-Forwards: 70
From: "1002 SIP" <sip:[email protected]>;tag=as030b8717
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 101 INVITE
User-Agent: Asterisk PBX 20.12.0
Date: Sat, 06 Dec 2025 00:13:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 1539663332 1539663332 IN IP4 10.0.0.1
s=Asterisk PBX 20.12.0
c=IN IP4 10.0.0.1
t=0 0
m=audio 10346 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

*Dec  6 00:13:46.267: //804/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6c9e3548;rport
From: "1002 SIP" <sip:[email protected]>;tag=as030b8717
To: <sip:[email protected]>
Date: Sat, 06 Dec 2025 00:13:46 GMT
Call-ID: [email protected]:5060
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-16.7.1
Session-ID: 00000000000000000000000000000000;remote=51cc615e84d35995ba7be4b648bb4268
Content-Length: 0


*Dec  6 00:13:46.293: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK250110F
Remote-Party-ID: "1002 SIP" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>
Date: Sat, 06 Dec 2025 00:13:46 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 1142279398-3513717232-2283246479-3520487954
User-Agent: Cisco-SIPGateway/IOS-16.7.1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1764980026
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-ID: 51cc615e84d35995ba7be4b648bb4268;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 6033 8900 IN IP4 10\.0\.0\.3
s=SIP Call
c=IN IP4 10.0.0.1
t=0 0
m=audio 10346 RTP/AVP 8 101
c=IN IP4 10.0.0.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

*Dec  6 00:13:46.314: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK250110F;received=10.0.0.3;rport=49594
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>;tag=as6519b84f
Call-ID: [email protected]
CSeq: 101 INVITE
Server: Asterisk PBX 20.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="530fcc33"
Content-Length: 0


*Dec  6 00:13:46.319: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK250110F
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>;tag=as6519b84f
Date: Sat, 06 Dec 2025 00:13:46 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Session-ID: 51cc615e84d35995ba7be4b648bb4268;remote=3020892098c357b1a089c1a66e441917
Content-Length: 0


*Dec  6 00:13:46.320: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK25113D8
Remote-Party-ID: "1002 SIP" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>
Date: Sat, 06 Dec 2025 00:13:46 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
Cisco-Guid: 1142279398-3513717232-2283246479-3520487954
User-Agent: Cisco-SIPGateway/IOS-16.7.1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1764980026
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Authorization: Digest username="9010",realm="asterisk",uri="sip:[email protected]:5060",response="b173bacae888b4b10bc5e875ebcdab76",nonce="530fcc33",algorithm=MD5
Max-Forwards: 69
Session-ID: 51cc615e84d35995ba7be4b648bb4268;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 6033 8900 IN IP4 10\.0\.0\.3
s=SIP Call
c=IN IP4 10.0.0.1
t=0 0
m=audio 10346 RTP/AVP 8 101
c=IN IP4 10.0.0.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

*Dec  6 00:13:46.348: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK25113D8;received=10.0.0.3;rport=49594
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 20.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


*Dec  6 00:13:46.349: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK25113D8;received=10.0.0.3;rport=49594
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>;tag=as3abf3ccf
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 20.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces,timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 230

v=0
o=root 190690963 190690963 IN IP4 10.0.0.1
s=Asterisk PBX 20.12.0
c=IN IP4 10.0.0.1
t=0 0
m=audio 18440 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:140
a=sendrecv

*Dec  6 00:13:46.359: //805/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.2.15:5060;branch=z9hG4bK2521B3A
From: "1002 SIP" <sip:[email protected]>;tag=F9ED96-A56
To: <sip:[email protected]>;tag=as3abf3ccf
Date: Sat, 06 Dec 2025 00:13:46 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Authorization: Digest username="9010",realm="asterisk",uri="sip:[email protected]:5060",response="b173bacae888b4b10bc5e875ebcdab76",nonce="530fcc33",algorithm=MD5
Allow-Events: telephone-event
Content-Length: 0


*Dec  6 00:13:46.367: //804/4415CCE68817/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK6c9e3548;rport
From: "1002 SIP" <sip:[email protected]>;tag=as030b8717
To: <sip:[email protected]>;tag=F9EDE2-1E06
Date: Sat, 06 Dec 2025 00:13:46 GMT
Call-ID: [email protected]:5060
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <sip:[email protected]>;party=called;screen=no;privacy=off
Contact: <sip:[email protected]:5060>
Supported: replaces
Server: Cisco-SIPGateway/IOS-16.7.1
Session-ID: eb3c59a243515ac3a570e1ac974b9441;remote=51cc615e84d35995ba7be4b648bb4268
Session-Expires:  1800;refresher=uas
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 236

v=0
o=CiscoSystemsSIP-GW-UserAgent 8767 5274 IN IP4 10.0.2.15
s=SIP Call
c=IN IP4 10.0.0.1
t=0 0
m=audio 18440 RTP/AVP 8 101
c=IN IP4 10.0.0.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

*Dec  6 00:13:46.389: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK2c09efe3;rport
Max-Forwards: 70
From: "1002 SIP" <sip:[email protected]>;tag=as030b8717
To: <sip:[email protected]>;tag=F9EDE2-1E06
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 101 ACK
User-Agent: Asterisk PBX 20.12.0
Content-Length: 0

Cube Config:
Code: 
Router#show run
Building configuration...

Current configuration : 14648 bytes
!
! Last configuration change at 00:56:22 UTC Sat Dec 6 2025
!
version 16.7
service timestamps debug datetime msec
service timestamps log datetime msec
service call-home
platform qfp utilization monitor load 80
no platform punt-keepalive disable-kernel-core
platform console serial
!
hostname Router
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
clock timezone UTC 1 0
call-home
 ! If contact email address in call-home is configured as [email protected]
 ! the email address configured in Cisco Smart License Portal will be used as contact email address to send SCH notifications.
 contact-email-addr [email protected]
 profile "CiscoTAC-1"
  active
  destination transport-method http
  no destination transport-method email
!
!
!
!
!
!
ip nbar http-services
!
!
!
ip host CSR1000V 10.0.2.15
ip name-server 8.8.8.8
ip domain name lab.local
!
!
!
!
!
!
!
!
!
!
subscriber templating
!
!
!
!
!
!
!
!
flow record defaultApplicationTraffic
 match ipv4 protocol
 match ipv4 source address
 match ipv4 destination address
 match transport source-port
 match transport destination-port
 collect transport tcp flags
 collect counter packets long
 collect timestamp sys-uptime first
 collect timestamp sys-uptime last
!
multilink bundle-name authenticated
!
!
!
!
!
crypto pki trustpoint SLA-TrustPoint
 enrollment terminal
 revocation-check crl
!
!
crypto pki certificate chain SLA-TrustPoint
 certificate ca 01
XXX
      quit
!
crypto pki certificate pool
 cabundle nvram:ios_core.p7b
!
!
!
!
voice service voip
 ip address trusted list
  ipv4 10.0.0.0 255.255.255.0
 rtp-port range 10000 20000
 mode border-element
 media flow-around
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 sip
  sip-profiles 2
!
!
voice class uri 1 sip
 host 10.0.0.3
voice class media 10
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
!
!
voice class sip-profiles 1
 request INVITE sip-header Contact modify "10\.0\.2\.15" "10\.0\.0\.3"
 response 200 sip-header Contact modify "10\.0\.2\.15" "10\.0\.0\.3"
 request INVITE sdp-header Connection-Info modify "header-value-regex" "c=IN"
 request INVITE sdp-header Session-Owner modify "10\.0\.2\.15" "10\.0\.0\.3"
 request INVITE sdp-header Connection-Info modify "10\.0\.2\.15" "10\.0\.0\.3"
 response 200 sdp-header Connection-Info modify "10\.0\.2\.15" "10\.0\.0\.3"
 request INVITE sdp-header Session-Owner modify "10\.0\.2\.15" "10\.0\.0\.3"
 request INVITE sdp-header Connection-Info modify "10\.0\.2\.15" "10\.0\.0\.3"
 response 200 sdp-header Connection-Info modify "10\.0\.2\.15" "10\.0\.0\.3"
!
voice class sip-profiles 10
 request INVITE sip-header From modify "sip:.*@" "sip:9010@"
 response 200 sdp-header Audio-Connection-Info modify "c=IN IP4 192.168.64.2" "c=IN IP4 ÖFFENTLICHEIP"
 response 180 sdp-header Audio-Connection-Info modify "c=IN IP4 192.168.64.2" "c=IN IP4 ÖFFENTLICHEIP"
 response 183 sdp-header Audio-Connection-Info modify "c=IN IP4 192.168.64.2" "c=IN IP4 ÖFFENTLICHEIP"
!
voice class sip-profiles 200
 request INVITE sip-header Contact modify "<sip:(.*)@(.*)>" "<sip:\[email protected]>"
!
voice class sip-profiles 20
!
voice class sip-profiles 30
!
voice class sip-profiles 100
 request INVITE sdp-header Connection-Info remove
 response 200 sdp-header Connection-Info remove
 request INVITE sdp-header Connection-Info remove
 request INVITE sdp-header Connection-Info modify "192.168.64.2" "ÖFFENTLICHEIP"
 response 200 sdp-header Connection-Info modify "192.168.64.2" "ÖFFENTLICHEIP"
 request INVITE sdp-header Connection-Info modify "192.168.64.2" "ÖFFENTLICHEIP"
 response 200 sdp-header Connection-Info modify "192.168.64.2" "ÖFFENTLICHEIP"
!
voice class sip-profiles 3
 request INVITE sip-header From modify "sip:.*@" "sip:9020@"
 request INVITE sip-header Contact modify "sip:.*@" "sip:9020@"
 request INVITE sip-header From modify "sip:.*@" "sip:9020@"
 request INVITE sip-header Contact modify "sip:.*@" "sip:9020@"
!
voice class sip-profiles 2
 request INVITE sip-header From modify "sip:.*@" "sip:9010@"
 request INVITE sip-header Contact modify "sip:.*@" "sip:9010@"
 request INVITE sdp-header Session-Owner modify "10\.0\.2\.15" "10\.0\.0\.3"
 request INVITE sdp-header Connection-Info modify "10\.0\.2\.15" "10\.0\.0\.3"
 response 200 sdp-header Connection-Info modify "10\.0\.2\.15" "10\.0\.0\.3"
!
!
voice class sip-copylist 100
 sip-header From
 sip-header To
!
voice class sip-copylist 1
 sip-header From
 sip-header To
!
voice class server-group 1
 ipv4 10.0.0.1
!
!
!
!
!
!
media profile nr 1
 noisefloor -40
!
media class 1
 nr profile 1
!
license udi pid CSR1000V sn 9Y1Y0T6X8U7
license smart enable
license smart privacy
diagnostic bootup level minimal
!
spanning-tree extend system-id
!
!
!
username user privilege 15 secret 5 XXX
username 9010 password 0 XXX realm asterisk
username 9020 password 0 XXX realm asterisk
!
redundancy
!
!
!
!
!
!
cdp run
!
class-map match-all WEBUI-MULTIMEDIA_CONFERENCING-DSCP
 match dscp af41
class-map match-all WEBUI-BROADCAST_VIDEO-NBAR
 match protocol attribute traffic-class broadcast-video
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-VOICE-NBAR
 match protocol attribute traffic-class voip-telephony
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-BULK_DATA-NBAR
 match protocol attribute traffic-class bulk-data
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-SIGNALING-NBAR
 match protocol attribute traffic-class signaling
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-NETWORK_CONTROL-DSCP
 match dscp cs6
class-map match-all WEBUI-SCAVENGER-NBAR
 match protocol attribute business-relevance business-irrelevant
class-map match-all WEBUI-SCAVENGER-DSCP
 match dscp cs1
class-map match-all WEBUI-NETWORK_CONTROL-NBAR
 match protocol attribute traffic-class network-control
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-SIGNALING-DSCP
 match dscp cs3
class-map match-all WEBUI-BULK_DATA-DSCP
 match dscp af11
class-map match-all WEBUI-BROADCAST_VIDEO-DSCP
 match dscp cs5
class-map match-all WEBUI-MULTIMEDIA_CONFERENCING-NBAR
 match protocol attribute traffic-class multimedia-conferencing
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-VOICE-DSCP
 match dscp ef
class-map match-all WEBUI-NETWORK_MANAGEMENT-NBAR
 match protocol attribute traffic-class ops-admin-mgmt
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-MULTIMEDIA_STREAMING-DSCP
 match dscp af31
class-map match-all WEBUI-REALTIME_INTERACTIVE-NBAR
 match protocol attribute traffic-class real-time-interactive
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-TRANSACTIONAL_DATA-DSCP
 match dscp af21
class-map match-all WEBUI-REALTIME_INTERACTIVE-DSCP
 match dscp cs4
class-map match-all WEBUI-TRANSACTIONAL_DATA-NBAR
 match protocol attribute traffic-class transactional-data
 match protocol attribute business-relevance business-relevant
class-map match-all WEBUI-NETWORK_MANAGEMENT-DSCP
 match dscp cs2
class-map match-all WEBUI-MULTIMEDIA_STREAMING-NBAR
 match protocol attribute traffic-class multimedia-streaming
 match protocol attribute business-relevance business-relevant
!
policy-map WEBUI-MARKING-IN
 class WEBUI-VOICE-NBAR
  set dscp ef
 class WEBUI-BROADCAST_VIDEO-NBAR
  set dscp cs5
 class WEBUI-REALTIME_INTERACTIVE-NBAR
  set dscp cs4
 class WEBUI-MULTIMEDIA_CONFERENCING-NBAR
  set dscp af41
 class WEBUI-MULTIMEDIA_STREAMING-NBAR
  set dscp af31
 class WEBUI-SIGNALING-NBAR
  set dscp cs3
 class WEBUI-NETWORK_CONTROL-NBAR
  set dscp cs6
 class WEBUI-NETWORK_MANAGEMENT-NBAR
  set dscp cs2
 class WEBUI-TRANSACTIONAL_DATA-NBAR
  set dscp af21
 class WEBUI-BULK_DATA-NBAR
  set dscp af11
 class WEBUI-SCAVENGER-NBAR
  set dscp cs1
 class class-default
  set dscp default
policy-map WEBUI-QUEUING-OUT
 class WEBUI-VOICE-DSCP
  priority percent 10
 class WEBUI-BROADCAST_VIDEO-DSCP
  priority percent 10
 class WEBUI-REALTIME_INTERACTIVE-DSCP
  priority percent 13
 class WEBUI-NETWORK_CONTROL-DSCP
  bandwidth percent 2
 class WEBUI-SIGNALING-DSCP
  bandwidth percent 2
 class WEBUI-NETWORK_MANAGEMENT-DSCP
  bandwidth percent 3
 class WEBUI-MULTIMEDIA_CONFERENCING-DSCP
  bandwidth percent 10
  fair-queue
  random-detect dscp-based
 class WEBUI-MULTIMEDIA_STREAMING-DSCP
  bandwidth percent 10
  fair-queue
  random-detect dscp-based
 class WEBUI-TRANSACTIONAL_DATA-DSCP
  bandwidth percent 10
  fair-queue
  random-detect dscp-based
 class WEBUI-BULK_DATA-DSCP
  bandwidth percent 4
  fair-queue
  random-detect dscp-based
 class WEBUI-SCAVENGER-DSCP
  bandwidth percent 1
 class class-default
  bandwidth percent 25
  fair-queue
  random-detect dscp-based
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
!
interface GigabitEthernet1
 description Router
 ip address 10.0.2.15 255.255.255.0
 negotiation auto
 no mop enabled
 no mop sysid
!
interface GigabitEthernet2
 description Local IP
 ip address 192.168.20.1 255.255.255.0
 ip nat inside
 ip nbar protocol-discovery
 negotiation auto
 no mop enabled
 no mop sysid
 spanning-tree portfast trunk
 service-policy input WEBUI-MARKING-IN
 service-policy output WEBUI-QUEUING-OUT
!
interface GigabitEthernet3
 description Public IP
 ip address 10.0.0.3 255.255.255.0
 shutdown
 negotiation auto
 no mop enabled
 no mop sysid
 spanning-tree portfast trunk
!
no ip nat service sip tcp port 5060
no ip nat service sip udp port 5060
ip nat inside source route-map track-primary-if interface GigabitEthernet3 overload
ip nat inside source list NAT_ACL interface GigabitEthernet1 overload
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip dns server view-group default
ip route 0.0.0.0 0.0.0.0 10.0.2.2
!
ip ssh version 2
ip ssh server algorithm mac hmac-sha2-256 hmac-sha2-512
ip ssh server algorithm encryption aes256-ctr aes192-ctr aes128-ctr
ip ssh server algorithm kex diffie-hellman-group14-sha1 diffie-hellman-group-exchange-sha1
ip ssh server algorithm authentication publickey
!
!
ip access-list standard NAT_ACL
 permit 192.168.20.0 0.0.0.255
!
!
route-map track-primary-if permit 1
 match ip address 197
 set interface GigabitEthernet3
!
!
!
control-plane
!
!
!
!
dial-peer voice 9020 voip
 destination-pattern 9020
 session protocol sipv2
 session target ipv4:192.168.178.31
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
!
dial-peer voice 5000 voip
 description VoIP zu Asterisk WG
 destination-pattern 5000
 progress_ind alert enable 8
 session protocol sipv2
 session target ipv4:10.0.0.1
 incoming called-number 5000
 voice-class sip profiles 2
 voice-class sip pass-thru content custom-sdp
 voice-class sip copy-list 100
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 9010 voip
 description VoIP zu Asterisk WG
 destination-pattern 5...
 progress_ind alert enable 8
 session protocol sipv2
 session target ipv4:10.0.0.1
 incoming called-number 5000
 voice-class sip profiles 1
 voice-class sip options-keepalive
 voice-class sip pass-thru content custom-sdp
 voice-class sip copy-list 100
 voice-class sip bind control source-interface GigabitEthernet1
 voice-class sip bind media source-interface GigabitEthernet1
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
!
sip-ua
 credentials username 9010 password 7 XXX realm ÖFFENTLICHEIP
 credentials username 9010 password 7 XXX realm lab.local
 credentials username 9010 password 7 XXX realm asterisk
 credentials username 1005 password 7 XXX realm asterisk
 authentication username 1005 password 7 XXX
 authentication username 1005 password 7 XXX realm asterisk
 retry invite 1
 retry register 10
!
!
line con 0
 stopbits 1
line vty 0 4
 login local
 transport input ssh
!
wsma agent exec
!
wsma agent config
!
wsma agent filesys
!
wsma agent notify
!
!
end

Router#

Kein RTP:
Code: 
Router#show call active voice brief
<ID>: <CallID> <start>ms.<index> (<start>) +<connect> pid:<peer_id> <dir> <addr> <state>
  dur hh:mm:ss tx:<packets>/<bytes> rx:<packets>/<bytes> dscp:<packets violation> media:<packets violation> audio tos:<audio tos value> video tos:<video tos value>
 IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms lost:<lost>/<early>/<late>
  delay:<last>/<min>/<max>ms <codec> <textrelay> <transcoded
 
 media inactive detected:<y/n> media cntrl rcvd:<y/n> timestamp:<time>
 
 long duration call detected:<y/n> long duration call duration :<sec> timestamp:<time>
 LostPacketRate:<%> OutOfOrderRate:<%>
 LocalUUID:<%> RemoteUUID:<%>
 VRF:<%>
  MODEMPASS <method> buf:<fills>/<drains> loss <overall%> <multipkt>/<corrected>
   last <buf event time>s dur:<Min>/<Max>s
 FR <protocol> [int dlci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
  <codec> (payload size)
 ATM <protocol> [int vpi/vci cid] vad:<y/n> dtmf:<y/n> seq:<y/n>
  <codec> (payload size)
 Tele <int> (callID) [channel_id] tx:<tot>/<v>/<fax>ms <codec> noise:<l> acom:<l> i/o:<l>/<l> dBm
  MODEMRELAY info:<rcvd>/<sent>/<resent> xid:<rcvd>/<sent> total:<rcvd>/<sent>/<drops>
         speeds(bps): local <rx>/<tx> remote <rx>/<tx>
 Proxy <ip>:<audio udp>,<video udp>,<tcp0>,<tcp1>,<tcp2>,<tcp3> endpt: <type>/<manf>
 bw: <req>/<act> codec: <audio>/<video>
  tx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
 rx: <audio pkts>/<audio bytes>,<video pkts>/<video bytes>,<t120 pkts>/<t120 bytes>
 


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
1A35 : 845 17186500ms.1 (*01:33:43.187 UTC Sat Dec 6 2025) +90 pid:5000 Answer 9010 active
 dur 00:00:23 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0x0 video tos:0x0
 IP 10.0.0.1:16788 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw 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:c12a02fc02fa5b44a00c7c13221f0ee6
 RemoteUUID:049677687e0e5cc5a9a0090ae9b63f7d
 VRF: NA
1A35 : 846 17186520ms.1 (*01:33:43.207 UTC Sat Dec 6 2025) +60 pid:5000 Originate 5000 active
 dur 00:00:23 tx:0/0 rx:0/0 dscp:0 media:0 audio tos:0x0 video tos:0x0
 IP 10.0.0.1:11220 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711alaw 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:1c9dfd28dccd58d7bcedfc5812a7395c
 RemoteUUID:c12a02fc02fa5b44a00c7c13221f0ee6
 VRF: NA

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

Router#
 
Zuletzt bearbeitet:
Ich habe das CUBE in das lokale Subnetz des Lancom Routers gebracht und im Lancom ein Site-to-site VPN auf Layer 2 zum entfernten Asterisk aufgebaut. Nun funktioniert im Asterisk Dial-Peer host=IPDESCUBES. Leider liegt der gesamte Audio-Leg bei Asterisk und nicht beim CUBE. MTP wäre wohl notwendig, schaffe es nur nicht. CUCM kann aktiv MTP konfigurieren, das weiß ich, nur gibt es bei Asterisk nicht eine Funktion die das gleiche bewirkt, sodass sie nur die SIP Signalisierung macht und dem CUBE das RTP überlässt? Die Option directmedia=yes nützt nichts, der DSP im CUBE wird nicht genutzt, was ja das eigentliche Ziel ist.
 
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