Accepted Solutions
Based on your provided information I would think that something along the way with this should do the trick for you.
** Inbound ** voice class sip-profiles 100 rule 5 request INVITE sip-header SIP-Req-URI modify "70.1.2.3" "10.21.10.12" rule 10 response ANY sdp-header Audio-Attribute modify "10.21.10.12" "70.1.2.3" rule 20 request ANY sdp-header Connection-Info modify "70.1.2.3" "10.21.10.12" rule 30 response ANY sdp-header Audio-Attribute modify "70.1.2.3" "10.21.10.12" rule 40 response ANY sdp-header Connection-Info modify "70.1.2.3" "10.21.10.12" rule 50 request ANY sdp-header mline-index 1 c= modify "70.1.2.3" "10.21.10.12" rule 60 response ANY sdp-header mline-index 1 c= modify "70.1.2.3" "10.21.10.12" rule 70 request ANY sdp-header Audio-Attribute modify "a=candidate:1 1 (.*) 70.1.2.3" "a=candidate:1 1 \1 10.21.10.12" rule 80 request ANY sdp-header Audio-Attribute modify "a=candidate:1 2 (.*) 70.1.2.3" "a=candidate:1 2 \1 10.21.10.12" rule 90 response ANY sdp-header Audio-Attribute modify "a=candidate:1 1 (.*) 70.1.2.3" "a=candidate:1 1 \1 10.21.10.12" rule 100 response ANY sdp-header Audio-Attribute modify "a=candidate:1 2 (.*) 70.1.2.3" "a=candidate:1 2 \1 10.21.10.12" rule 110 request ANY sdp-header Audio-Attribute modify "10.21.10.12" "70.1.2.3" ** Outbound ** voice class sip-profiles 200 rule 10 response 200 sdp-header Audio-Connection-Info modify "0.0.0.0" "70.1.2.3" rule 20 response ANY sdp-header Connection-Info modify "10.21.10.12" "70.1.2.3" rule 30 response ANY sdp-header Audio-Connection-Info modify "10.21.10.12" "70.1.2.3" rule 40 request ANY sdp-header Connection-Info modify "10.21.10.12" "70.1.2.3" rule 50 request ANY sdp-header Audio-Connection-Info modify "10.21.10.12" "70.1.2.3" rule 60 response ANY sdp-header Audio-Attribute modify "a=rtcp:(.) 10.21.10.12" "a=rtcp:\1 70.1.2.3 " rule 70 request ANY sdp-header Audio-Attribute modify "a=rtcp:(.) 10.21.10.12" "a=rtcp:\1 70.1.2.3 " rule 80 response ANY sdp-header Audio-Attribute modify "a=candidate:1 1(.) 10.21.10.12 (.) " "a=candidate:1 1\1 70.1.2.3 \2" rule 90 request ANY sdp-header Audio-Attribute modify "a=candidate:1 1(.) 10.21.10.12 (.) " "a=candidate:1 1\1 70.1.2.3 \2" rule 100 response ANY sdp-header Audio-Attribute modify "a=candidate:1 2(.) 10.21.10.12 (.) " "a=candidate:1 2\1 70.1.2.3 \2" rule 110 request ANY sdp-header Audio-Attribute modify "a=candidate:1 2(.) 10.21.10.12 (.) " "a=candidate:1 2\1 70.1.2.3 \2" ** SIP option ping ** voice class sip-profiles 299 rule 5 request ANY sip-header Via modify "SIP(.*) 10.21.10.12(.*)" "SIP\1 70.1.2.3\2" rule 10 request OPTIONS sip-header From modify "<sip:10.21.10.12" "<sip:<FQDN name of the router>" rule 20 request OPTIONS sip-header Contact modify "<sip:10.21.10.12" "<sip:<FQDN name of the router>" rule 30 response ANY sdp-header Connection-Info modify "10.21.10.12" "70.1.2.3" rule 40 response ANY sdp-header Audio-Connection-Info modify "10.21.10.12" "70.1.2.3"
Once you're happy with the SIP profiles and have tested them in the tool that I linked to in another answer then you set it to use on the dial peers with this.
voice class sip-options-keepalive 200 sip-profiles 299 ! voice service voip sip sip-profiles 100 inbound ! dial-peer voice 100 voip !(If this is your inbound dial peer from ITSP) voice-class sip profiles 100 inbound ! dial-peer voice 200 voip !(If this is your outbound dial peer to ITSP) voice-class sip profiles 200
voice-class sip options-keepalive profile 200
Most of this is based on how to handle a setup with NAT for Direct Routing with the MS Teams calling platform, but it should be applicable for the most for anything that does NAT.
For detailed information on how this works in IOS/IOS-XE I'm going to reference my likely most linked to document these days.
Can you share the full information of the invite from debug ccsip message and also the details of the public and private IP that the NAT is setup for? If you do not want to share your actual public IP just change this information to something else, but keep it consistent.
What you need is to have a SIP profile that modify the IP address information from the public IP to the private IP in the SIP dialogue so that the router recognises the invite as destined to it.
I think that the command on the dial-peer would be "voice-class sip profiles 100 inbound".
Apart from this there would very likely be a few more headers that needs to be modified and quite likely it also needs to be done for outbound calls. Best is to look at the complete dialogue for a call attempt and then iterate through the entire thing to outline what is needed to be modified with a SIP profile. Once that is done I would strongly recommend to use the excellent tool on https://cway.cisco.com/tools/SipProfileTest/ to test out that the SIP profile does all the needed changes to the SIP headers before putting it into the gateway.
Based on your provided information I would think that something along the way with this should do the trick for you.
** Inbound ** voice class sip-profiles 100 rule 5 request INVITE sip-header SIP-Req-URI modify "70.1.2.3" "10.21.10.12" rule 10 response ANY sdp-header Audio-Attribute modify "10.21.10.12" "70.1.2.3" rule 20 request ANY sdp-header Connection-Info modify "70.1.2.3" "10.21.10.12" rule 30 response ANY sdp-header Audio-Attribute modify "70.1.2.3" "10.21.10.12" rule 40 response ANY sdp-header Connection-Info modify "70.1.2.3" "10.21.10.12" rule 50 request ANY sdp-header mline-index 1 c= modify "70.1.2.3" "10.21.10.12" rule 60 response ANY sdp-header mline-index 1 c= modify "70.1.2.3" "10.21.10.12" rule 70 request ANY sdp-header Audio-Attribute modify "a=candidate:1 1 (.*) 70.1.2.3" "a=candidate:1 1 \1 10.21.10.12" rule 80 request ANY sdp-header Audio-Attribute modify "a=candidate:1 2 (.*) 70.1.2.3" "a=candidate:1 2 \1 10.21.10.12" rule 90 response ANY sdp-header Audio-Attribute modify "a=candidate:1 1 (.*) 70.1.2.3" "a=candidate:1 1 \1 10.21.10.12" rule 100 response ANY sdp-header Audio-Attribute modify "a=candidate:1 2 (.*) 70.1.2.3" "a=candidate:1 2 \1 10.21.10.12" rule 110 request ANY sdp-header Audio-Attribute modify "10.21.10.12" "70.1.2.3" ** Outbound ** voice class sip-profiles 200 rule 10 response 200 sdp-header Audio-Connection-Info modify "0.0.0.0" "70.1.2.3" rule 20 response ANY sdp-header Connection-Info modify "10.21.10.12" "70.1.2.3" rule 30 response ANY sdp-header Audio-Connection-Info modify "10.21.10.12" "70.1.2.3" rule 40 request ANY sdp-header Connection-Info modify "10.21.10.12" "70.1.2.3" rule 50 request ANY sdp-header Audio-Connection-Info modify "10.21.10.12" "70.1.2.3" rule 60 response ANY sdp-header Audio-Attribute modify "a=rtcp:(.) 10.21.10.12" "a=rtcp:\1 70.1.2.3 " rule 70 request ANY sdp-header Audio-Attribute modify "a=rtcp:(.) 10.21.10.12" "a=rtcp:\1 70.1.2.3 " rule 80 response ANY sdp-header Audio-Attribute modify "a=candidate:1 1(.) 10.21.10.12 (.) " "a=candidate:1 1\1 70.1.2.3 \2" rule 90 request ANY sdp-header Audio-Attribute modify "a=candidate:1 1(.) 10.21.10.12 (.) " "a=candidate:1 1\1 70.1.2.3 \2" rule 100 response ANY sdp-header Audio-Attribute modify "a=candidate:1 2(.) 10.21.10.12 (.) " "a=candidate:1 2\1 70.1.2.3 \2" rule 110 request ANY sdp-header Audio-Attribute modify "a=candidate:1 2(.) 10.21.10.12 (.) " "a=candidate:1 2\1 70.1.2.3 \2" ** SIP option ping ** voice class sip-profiles 299 rule 5 request ANY sip-header Via modify "SIP(.*) 10.21.10.12(.*)" "SIP\1 70.1.2.3\2" rule 10 request OPTIONS sip-header From modify "<sip:10.21.10.12" "<sip:<FQDN name of the router>" rule 20 request OPTIONS sip-header Contact modify "<sip:10.21.10.12" "<sip:<FQDN name of the router>" rule 30 response ANY sdp-header Connection-Info modify "10.21.10.12" "70.1.2.3" rule 40 response ANY sdp-header Audio-Connection-Info modify "10.21.10.12" "70.1.2.3"
Once you're happy with the SIP profiles and have tested them in the tool that I linked to in another answer then you set it to use on the dial peers with this.
voice class sip-options-keepalive 200 sip-profiles 299 ! voice service voip sip sip-profiles 100 inbound ! dial-peer voice 100 voip !(If this is your inbound dial peer from ITSP) voice-class sip profiles 100 inbound ! dial-peer voice 200 voip !(If this is your outbound dial peer to ITSP) voice-class sip profiles 200
voice-class sip options-keepalive profile 200
Most of this is based on how to handle a setup with NAT for Direct Routing with the MS Teams calling platform, but it should be applicable for the most for anything that does NAT.
For detailed information on how this works in IOS/IOS-XE I'm going to reference my likely most linked to document these days.
