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Changing the AS Bandwidth allocation for outgoing call invites in SDP

stuart.pannell
Beginner
Beginner

We have a ITSP trying to implement VoLTE on their mobile network and are coming up with some voice quality issues calling in to Cisco CUCM 12.0 via an Audiocodes SBC. We are using standard G711 codecs on the SIP trunk to the SBC and the SDP offer has the standard bandwidth offerings of 64k for G711. Our Service providers VoLTE deployment is using an AS of 80k bandwidth for PCM 8 calls. They are dropping packets between the SBC network and the Mobile LTE network yet have asked us if we can change our bit rate from 64k to 80k to match theirs.

 

Is there a way of changing this on the outgoing SIP trunk?

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