I am trying to setup a call recording via a cisco unified callmager (cucm), asterisk and a softphone client. The SIP signaling appears to be correct, but asterisk does not receive any rtp packets from the softphone client.
The Setup in more detail:
The CUCM registers a sip trunk. On inbound calls on the sip trunk, the cucm sends a variety of invites to establish a two way connection between the softphone client and the sip trunk with asterisk as recording participant:
- CUCM invites asterisk in sendonly mode; tells asterisk the media address of the softphone. Asterisk accepts the invite as expected (twice a call for two separate udp ports)
- CUCM invites the softphone client and tells the client the media address of the sip trunk
- CUCM invites the softphone client and tells the client the media address of asterisk in receiveonly mode (twice a call for the two different udp ports from 1.)
- CUCM invites the softphone client without a media address in inactive mode (I assume to enable future sip signaling)
The asterisk console logs and wireshark captures indicate, that the sip signaling (media addresses and ports) is correct. The sip trunk caller can also talk to the softphone client, proving that the rtp stream between the sip trunk and the softphone client is setup correctly (wireshark capture also shows inbound and outgoing rtp packets between the softphone client and the sip trunk).
The network capture also shows outgoing rtp packets from the softphone client towards asterisk. Wireshark captures show that these packets also reach the asterisk machine on the port specified in the sdp but are not received by asterisk. The recording contains only silence and the enabled rtp debug shows no incoming packets in the asterisk console.
The asterisk recording dialplan consists of four statements; answer, start mixmonitor, wait till the call is finished, hangup. The asterisk log shows that these steps are executed as expected. the recording is started and finished accordingly but contains only silence.
A control routing directly between asterisk and the cucm (without the softphone client) is working as expected and recording successfully.
Further analysis with netstat on the asterisk machine indicates, that asterisk is not listening on the udp ports from the sdp, even though the invite was successful and not rejected. When using the direct routing without softphone the udp ports are listed in the netstat output.
Without any error message it is very difficult to figure out why asterisk is not listening on the udp ports / not receiving the rtp packets. Therefor i would be very thankful for any suggestion or advice on potential issues or where to investigate further.