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      G722 (HDVoice) is a new Codec that is supported by most current VOIP Phones, but NOT by most VOIP Providers.

      When I enable G722 on my system and make a call using a phone which supports G722, FreePBX places all calls in G722 mode, regardless of whether they are internal or external calls. For external calls, this means that Asterisk must transcode the call from G722 to G711, which introduces another latency step.

      I'd prefer to have some way to set the Codec by route, i.e. so that when I make internal calls, the calls use G722, but when I make an external call using a SIP Trunk, the call goes out on G711 (avoiding the transcoding/latency issues).

      I suggest that FreePBX add a feature that allows a user to select an internal vs. external Codec, or to set an outbound route Codec that will override the general Asterisk SIP codecs.

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            xrobau Rob Thomas added a comment -

            Confirmed that this also doesn't work in Asterisk 1.8, so yeah. Asterisk issue.

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            xrobau Rob Thomas added a comment - Confirmed that this also doesn't work in Asterisk 1.8, so yeah. Asterisk issue.
            Hide
            martin_anderson Martin Anderson added a comment -

            Just tested with FreePBX 12 and Asterisk 11, and it still locally bridges.

            Isn't Asterisk ALWAYS going to locally bridge if you support call recording, transfer, etc?

            Show
            martin_anderson Martin Anderson added a comment - Just tested with FreePBX 12 and Asterisk 11, and it still locally bridges. Isn't Asterisk ALWAYS going to locally bridge if you support call recording, transfer, etc?
            Hide
            xrobau Rob Thomas added a comment -

            It SHOULD be bridging, which means it doesn't need to transcode. Have a look at 'core show channel ' and then push tab to pick the two channels that are in use, and see if you can see why it's transcoding. It shouldn't be (and it didn't for me).

            Show
            xrobau Rob Thomas added a comment - It SHOULD be bridging, which means it doesn't need to transcode. Have a look at 'core show channel ' and then push tab to pick the two channels that are in use, and see if you can see why it's transcoding. It shouldn't be (and it didn't for me).
            Hide
            martin_anderson Martin Anderson added a comment -

            Did you test it on Asterisk 11, or 13? You previously said 13. I'm using 11.

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            martin_anderson Martin Anderson added a comment - Did you test it on Asterisk 11, or 13? You previously said 13. I'm using 11.
            Hide
            martin_anderson Martin Anderson added a comment -

            Just to clarify, I was using 1.8 until last week. I just installed the latest Distro with Asterisk 11 to see if it still had the same problem, and it does.

            I haven't tried 13.

            Show
            martin_anderson Martin Anderson added a comment - Just to clarify, I was using 1.8 until last week. I just installed the latest Distro with Asterisk 11 to see if it still had the same problem, and it does. I haven't tried 13.

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              • Assignee:
                xrobau Rob Thomas
                Reporter:
                AdHominem Martin Anderson
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