SIP Devices - One Way Audio Issue

[bptbm]

I have a new install and when implementing a UC360 device, I get audio when dialing in-house, but I am getting 1-way audio when going over SIP Trunks. When a 53XX phone dials out on the SIP Trunks, the call has 2-way audio, but not when using a Generic SIP Device. Also, when attempting to use a resilient node used as a Disaster Recovery site (each Controller has it's own SIP Trunks - Main site has 192 SIP Trunks, while Resilient Node has 144 SIP Trunks - COX Communications (SIP Carrier) - when we unplug either the SIP Trunks at the Edgemark device, or turn off the Virtual MCD to make the phones resilient to the DR site, we also are getting 1-way audio when testing the resiliency of the network...

Does anyone have an idea, as to what should be set for either the UC360 1-way audio or the Failover/Resilient SIP Trunks getting 1-way audio?

 

[sarond]

Is the uc360 getting an IP from network as the 53xx phones.
The UC360 won't use Option 125 for DHCP.

Which way does the audio fail? Transmit or receive from UC360 perspective.
Sounds like a routing issue.

 

[bptbm]

Come to find out the SIP vendor didn't have all the IP Networks for the various sites in it's Edgemark device. Once they added the various IP Schemes into it's database, the one-way audio issue went away...Thanks for any help you guys and gals provided.

 

[GZgidnick]

I guess you have VLANs. Check that the UC360 is going onto the correct VLAN.
Another thing to keep in mind, if you use a camera for video conferencing, the UC360 and the Camera need to be on the same VLAN. You would usually put them on the data VLAN, but allow the inter vlan routing to and from the 3300.

Now, you can check in which VLAN that UC360 from, I believe the 'SIP Settings" section or "Network" section. You can also manually adjust the VLAN tag there as well.
Or you can use the 3300 'Maintenance and Diagnostic' , than select 'All IP Phones' form. You can search by MAC address and find the subnet assigned to the UC360.

Hate when sound overconfident, but one-way audio is 99.999% voice payload routing issue.
The fact that your signaling messages, that is SIP, reach each destination and successfully establish the session, doesn't mean that your RTP packets will do the same.
SIP is signaling protocol, hence intelligent.
RTP although layer 7 protocol, entirely depends on its Layer 4 UDP/TCP, therefore Layer 3 IP routed protocol to get to its destination. The fundamental network rules, such as default gateway, network routes etc apply here, which can take your payload to entirely different destination, hence one way audio.