Where do I make it a tip? Is there a category for that?

By the way, thank you James for your assistance, as well.

No, Rauland is 20ms. Turns out we needed to set $session to SDP and sdp.audio.a.1 to ptime:20. In case it helps anyone, our deploy plan wound up being: To=sip:%1*%2*%3@ $auth=false &net.sip.replacesdp.multipart=true $session=sdp &sdp.audio.a.1=ptime:20 This inserted the asterisks where we needed ...

Avaya was set to 20ms. We ran the call trace from the PBX.

PBX Vendor did a trace in the PBX that showed the Codec selection details.

The PBX is an Avaya CS1000E (Formerly Nortel CS1000E). The sets are Avaya 6120 (Nortel 6120), and Avaya 1120E (Nortel 1120E). It is a standalone Siginalling server installed with the SIP gateway application. They are not registering anything through NRS, so the Avaya (Nortel) Sip gateway is pointed ...

We have RTP Relay set to ON. We are attempting to call back the Rauland end points (staff station, bed station) from Avaya 6120 handhelds and Avaya 1120 desk phones. The Rauland end points acknowledge the call, but do not establish the audio connection between the station and Avaya devices. We have ...

I was wondering if the poster had resolved this issue. We are currently experiencing the same issue except that we are not getting two-way audio on any of the calls from the hand sets to the system. We are able to establish two-way audio both to the Nortel phones and the Rauland end points from X ...