Details

    • Type: Feature Requests
    • Status: Open
    • Priority: Trivial
    • Resolution: Unresolved
    • Affects Version/s: 2.9
    • Fix Version/s: 12
    • Component/s: Core
    • ToDo:
    • Target Release:
      PW

      Description

      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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            Hide
            tm1000 Andrew Nagy added a comment -

            All of this could be contained into one comment. I realize you are excited but for consistencies sake please don't post multiple times. Thank you.

            This ticket is currently not being worked on at this time.

            Show
            tm1000 Andrew Nagy added a comment - All of this could be contained into one comment. I realize you are excited but for consistencies sake please don't post multiple times. Thank you. This ticket is currently not being worked on at this time.
            Hide
            jmpg José Gonçalves added a comment - - edited

            This is a feature that I would also like to be available in FreePBX.
            I also would like to have a way to avoid transcoding for trunks that do not support G.722, while retaining the G.722 usage for internal calls.
            I'm currently running FreePBX 12 and Asterisk 11 over a Debian 8 distro.

            Show
            jmpg José Gonçalves added a comment - - edited This is a feature that I would also like to be available in FreePBX. I also would like to have a way to avoid transcoding for trunks that do not support G.722, while retaining the G.722 usage for internal calls. I'm currently running FreePBX 12 and Asterisk 11 over a Debian 8 distro.
            Hide
            tm1000 Andrew Nagy added a comment -

            We have no plans to add this feature at this time. Rob has confirmed to me it works as he has stated. I am putting this ticket to PW, meaning patches welcome, for anyone who wishes to do more work on it. We just do not have the time or resources.

            Show
            tm1000 Andrew Nagy added a comment - We have no plans to add this feature at this time. Rob has confirmed to me it works as he has stated. I am putting this ticket to PW, meaning patches welcome, for anyone who wishes to do more work on it. We just do not have the time or resources.
            Hide
            martin_anderson Martin Anderson added a comment - - edited

            Hi Andrew,

            I'm still a bit confused. This is a feature request, and so stating that it "works" has no meaning. The feature that I requested, having the ability to select codecs and priorities in the outbound route, is absent. Since the feature is not present, it is impossible for the presently non-existent feature to "work."

            I understand that Rob has said various things about how FreePBX and Asterisk work, but I couldn't fully understand what he meant. Did he write that on outbound calls, the codec that is available on all segments of the call is selected for the entire call with Asterisk 13, but not with Asterisk 1.8?

            Show
            martin_anderson Martin Anderson added a comment - - edited Hi Andrew, I'm still a bit confused. This is a feature request, and so stating that it "works" has no meaning. The feature that I requested, having the ability to select codecs and priorities in the outbound route, is absent. Since the feature is not present, it is impossible for the presently non-existent feature to "work." I understand that Rob has said various things about how FreePBX and Asterisk work, but I couldn't fully understand what he meant. Did he write that on outbound calls, the codec that is available on all segments of the call is selected for the entire call with Asterisk 13, but not with Asterisk 1.8?
            Hide
            JayG30 jayg30 added a comment -

            I've had this problem forever. I know for a fact I've tested all releases since Asterisk 11 through various FreePBX releases and the problem has always existed. Outbound calls to a SIP Trunking provider always results in transcoding from G.722 to G.711 if G.722 is listed as the first codec. Because of this I can't make use of HD audio on any of our phones at work. Very disappointing.

            Show
            JayG30 jayg30 added a comment - I've had this problem forever. I know for a fact I've tested all releases since Asterisk 11 through various FreePBX releases and the problem has always existed. Outbound calls to a SIP Trunking provider always results in transcoding from G.722 to G.711 if G.722 is listed as the first codec. Because of this I can't make use of HD audio on any of our phones at work. Very disappointing.

              People

              • Assignee:
                Unassigned
                Reporter:
                AdHominem Martin Anderson
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                Dates

                • Created:
                  Updated: