11-21-2022 08:30 PM - edited 11-21-2022 08:40 PM
Hi Tech People ,
we are trying to record a MGCP Audio Stream from A third Party Voice Recorder. we have VG400 integrated with CUCM via MGCP Protocol for PSTN Calling. we have no problem in recording the SIP and SCCP streams on same voice recorder. we can see the Communication between CUCM and Voice Gateway via MGCP using 2427 port from the following capture but Voice Recorder is unable to record a audio conversation.
Signal | 26 | 26 | 172.16.17.5:2427 (CUCM) | 172.16.36.5:2427 | 21:32:37 | |||
RTP 1 | 25116 | 24900 | 172.16.17.5:2427 (CUCM) | 172.16.36.5:2427 | 00:00:04 | 48 | 34343935 | 12320 |
RTP 2 | 140 | 140 | 172.16.36.5:2427 (VG400) | 172.16.52.5:2427 | 21:59:39 | 84 | 31333935 | 18009 |
RTP 3 | 0 | 0 | 0.0.0.0:0 | 0.0.0.0:0 | ||||
RTP 4 | 0 | 0 | 0.0.0.0:0 | 0.0.0.0:0 |
VG400 MGCP config
mgcp call-agent Site 1-CUCM 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface GigabitEthernet0/0/0
mgcp bind media source-interface GigabitEthernet0/0/0
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
ccm-manager mgcp
Topology is already attached. can any body share his experience if face the same troubleshooting.
11-22-2022 01:49 AM
Between which entities does the call flow?
Between an external caller and the analog phone on the VG port? (in your drawing between analog phone 1 and 2?). If yes, then the media won't go through the switch-port E0/0 and hence nothing will be recorded.
Media is (per default) flowing directly between the endpoints.
11-22-2022 04:42 AM
Hi b,winter,
Thanks for your reply.
The End point is connected to an FXS port on VG400. So Voice Recorder will take the Capture from the Network via SPAN protocol. I used the VG400 Ethernet address (MAC address) on voice Recorder to send the capture to voice Recorder. we can send the Stream through the IP address also. In short this is a local SPAN, Source and destination are on the switch.
11-22-2022 05:03 AM
Yeah, I got already what you wrote in your original post. But you didn't answer my question(s) ...
You only capture the traffic on a single LAN port (E0/0 in your drawing). So if the media packets aren't going through that port, you won't record anything.
If analoge phone 1 (connected to VG on FXS-port) is talking to analog phone 2 (someone external on FXO-port), the media is flowing between the FXS- and the FXO-port (See red line in the picture). No RTP packets will ever reach the LAN port of the VG, that you are recording with your SPAN-port. => Therefore, you don't see anything.
Only if the analog phone 1 or analog phone 2 is talking to someone internal (e.g. a normal Cisco phone in the LAN), then the media packets flow between the FXS-port and the internal party (See blue line in the picture) => therefore through the LAN port of the VG => therefore you can capture something.
11-22-2022 05:13 AM
Hi b,winter,
Thanks for your reply. i got your point. actully the ports of VG400 (FXS, FXO )are configured in Call Manager via MGCP. what about Call Manager function like you can see the following wireshark capture: The source is call manager and destination is VG400 and vice versa. This communication is going through the LAN. Correct me if misunderstand this.
Thanks in advance
Signal | 26 | 26 | 172.16.17.5:2427 (CUCM) | 172.16.36.5:2427 | 21:32:37 | |||
RTP 1 | 25116 | 24900 | 172.16.17.5:2427 (CUCM) | 172.16.36.5:2427 | 00:00:04 | 48 | 34343935 | 12320 |
RTP 2 | 140 | 140 | 172.16.36.5:2427 (VG400) | 172.16.52.5:2427 | 21:59:39 | 84 | 31333935 | 18009 |
RTP 3 | 0 | 0 | 0.0.0.0:0 | 0.0.0.0:0 | ||||
RTP 4 | 0 | 0 | 0.0.0.0:0 | 0.0.0.0:0 |
11-22-2022 05:27 AM - edited 11-22-2022 05:28 AM
Yes and no.
For signalling, CUCM will always be in the "path".
For media (voice/video RTP packets): per default, CUCM is never in the "path". Media always flows directly between the endpoints.
Found this schema drawing (don't get confused by sRTP and E1, the principle is important):
This will always be the same.
Doesn't matter if there is a phone or a gateway, or CUBE.
Doesn't matter if you use SIP, H.323, MGCP, SCCP.
And in principle, your wireshark confirms this, as you only see packets between CUCM and VG on port 2427 (MGCP signalling port).
Signal | 26 | 26 | 172.16.17.5:2427 (CUCM) | 172.16.36.5:2427 | 21:32:37 | |||
RTP 1 | 25116 | 24900 | 172.16.17.5:2427 (CUCM) | 172.16.36.5:2427 | 00:00:04 | 48 | 34343935 | 12320 |
RTP 2 | 140 | 140 | 172.16.36.5:2427 (VG400) | 172.16.52.5:2427 | 21:59:39 | 84 | 31333935 | 18009 |
Therefore, depending on which entity talks to which other entity, media will flow through the monitored LAN port of the VG or not.
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