Audio codec question

[JaskHill]

Hi, I have an audio codec question. It may not happen in real life. I am just curious.

Let's say, I register my SIP account to VSP (Voice-Service-Provider) A, in my OBI box.
    * VSP A supports codec G.711. It may support a few more codecs or may not.
    * I only enable G.711 in my OBI box. I disable all other codecs in my OBI box.

Then, my friend registers his SIP account to VSP B, in his OBI box (or any SIP device/soft-phone).
    * VSP B supports codec G.729. It may support a few more codecs or may not.
    * He only enables G.729 in his OBI box. He disables all other codecs in his OBI box.

If I call my friend, will the call go through?

Essentially, if two parties set totally different audio codec from each side ( but such codec is supported by their own VSP. But may not be supported by the other VSP ). Can they still call each other?

Thank you!

[drgeoff]

G.711 is the lingua franca of telcos.  They are all expected to support it, transcoding where necessary.  That routinely happens on calls between POTS and cellular.

If the endpoints are being called by their NANP numbers it is highly likely that the route between the two ITSPs will be over G.711 thus negating any quality advantage of G.729.

[JaskHill]

I understand it. I just happen to use G.711/729 as example here.

Basically, my question is: If person 1 is only using Codec A in his OBI, and person 2 is only using Codec B in his OBI, can person 1 & 2 still call each other, using SIP URL or whatever?

( Person 1 & 2 each registers to a different VSP ).

[drgeoff]

I understand it. I just happen to use G.711/729 as example here.

Basically, my question is: If person 1 is only using Codec A in his OBI, and person 2 is only using Codec B in his OBI, can person 1 & 2 still call each other, using SIP URL or whatever?

( Person 1 & 2 each registers to a different VSP ).

What SIP URL?

If you mean calling directly to the SIP URL (like sip:jask@jaskhill.com) of the other OBi (ie not going through ITSPs) I would expect the call to fail.

If you mean using sip:jaskhillsid@someitsp.com, the ITSPs may be clever enough to do the required transcoding.

But why? Just include G.711 in the OBis' list of codecs.

[JaskHill]

Thanks for the quick answer. Yes, that answer is what I am looking for, especailly the following piece:

        ...... the ITSPs may be clever enough to do the required transcoding.