[cisco-voip] SIP Load and Re-Invite

[Jason Aarons (AM)]

Is it normal for sip providers (say Verizon) to want to change codec mid-call or require your equipment can do it?  I understand CallManager 8.5 / SIP 7945 can't do this.  Is it a CallManager limitation or a phone load limitation or both for reinvite to change codec mid-call?

I'm a fan of codecs that dynamically change bandwidth (Silk, iLBC) but not G.722 to G711 to G.729 if the call degrades, but I guess with SIP you could possible do this based on the spec for re-invite. Or switch from G.729 to G.711 for faxes, etc.

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