cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
277
Views
0
Helpful
0
Replies

Changing the AS Bandwidth allocation for outgoing call invites in SDP

stuart.pannell
Level 1
Level 1

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?

0 Replies 0
Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: