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?
The Polycom video codec has an audio meter function that can measure the audio from the far side as well as the mic audio. When I send a test tone from an IP phone which is a 1004hz tone at -15 the Polycom sees this coming in at +18. Youch! I'm surprised it works as well as it does.
I am working with Polycom on this, but they don't seem to think it is their issue. Anybody out there working with Polycom codecs who has seen this?
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