I am using an H225 trunk (CCM 4.1.3) to integrate a Polycom video system. The call routing works fine, but the receive audio level on the Polycom side is too high, with distortion when someone speaks loudly. The calls are routed to/from the VSX7000 codecs by the Polycom SE200 gatekeeper. The voice RTP stream is direct between the Cisco phone (for an audio only call) and the Polycom VSX7000. I also have an ISDN PRI between the 2 systems, and had to pad down the audio out on the Cisco side by 9db for calls routed this way.
Any fix to this problem would likely require a code change to the Polycom codecs (or a global Cisco codec change), so I am not expecting any solutions here, just trying to understand this better.
So I have 3 questions:
1. Is there a standard RTP G711 normalized audio volume level?
2. How can I verify that the level from the Cisco side is correct?
3. Is it common to have RTP audio level issues between endpoints from different manufacturers?
Thanks!