Problem no recording when agent using Jabber VPN

[ibasuki]

Problem no recording when agent using Jabber VPN

Currently I'm facing issue where if CCX agent using VPN connection to office, recording is listed, but audio is silent.

After deep troubleshooting by collecting several wiresharks and log, this is our found:

• If agents are using LAN connection, calling and called endpoint's audio is present, recording is normal
• If agents are using VPN connection, calling and called endpoint's audio is present, but recording is silent.
• From agent with VPN connection, once call is answered, it creates 2 more RTP stream (total 3 RTP streams):

1. RTP to VG as PSTN endpoint

2. RTP to CUCM for near-end audio

3. RTP to CUCM for far-end audio

• From wireshark of that agent, I can hear RTP stream audio, that means Jabber on agent's side already sending RTP stream to CUCM.
• From CUCM trace that was collected, there is no RTP packet received by CUCM.

I also have consulted with Firewall Administrator. But on his side, he make sure no blocking between VPN Network to CUCM cluster (CUCM, Recording server, and VG are on same subnet). We also tried to disable SIP ALG on ASA, but still no luck.

Has anyone experience this before? Firewall is ASA+SFR version 6.6.4

Thank you.

[b.winter]

Which recording option have you configured? https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/configExamples/cucm_b_recording-use-cases.html#reference_1B976AEA3D7A570FA0F4B4526C5FE597

Are you using BiB recording?

Where in your RTP flow is the media flow to the recording server? Maybe the flow from the phone to the recording server is blocked on your VPN FW.
And why should there be a media flow to / from CUCM?

[Jonathan Schulenberg]

SIP ALG will almost certainly break it; ensure that is OFF.

And just to clarify: the RTP traffic should not be going to CUCM. Check your PCAP; it had better be going directly to the recording server IP address. If you see the packets going to CUCM you have an MTP involved that shouldn’t be there; figure out why (eg MTP Required checked) and disable/fix that.

Also verify that the codec negotiated for the call is supported by your recording server. It should have rejected the SIP INVITE if it didn’t like the SDP though.