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      13
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      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
          mickecarlsson added a comment -

          You can't set codecs on an Outbound Route.

          You can only set codecs on these places:

          • Endpoints, i.e. the phone itself, and the order to be used.
          • Extensions page in FreePBX. disallow and allow section.
          • Trunk Settings, PEER Details
          • Globally in Asterisk SIP Settings module.

          If you set the Endpoint to use G722 as a default codec, that is whats get sent to Asterisk to be used. If your calls go out on a trunk that does not have this coded Asterisk will transcode.

          Show
          mickecarlsson added a comment - You can't set codecs on an Outbound Route. You can only set codecs on these places: Endpoints, i.e. the phone itself, and the order to be used. Extensions page in FreePBX. disallow and allow section. Trunk Settings, PEER Details Globally in Asterisk SIP Settings module. If you set the Endpoint to use G722 as a default codec, that is whats get sent to Asterisk to be used. If your calls go out on a trunk that does not have this coded Asterisk will transcode.
          Hide
          mickecarlsson added a comment -

          Formatting was lost:

          • Endpoints, i.e. the phone itself, and the order to be used.
          • Extensions page in FreePBX. disallow and allow section.
          • Trunk Settings, PEER Details
          • Globally in Asterisk SIP Settings module.
          Show
          mickecarlsson added a comment - Formatting was lost: Endpoints, i.e. the phone itself, and the order to be used. Extensions page in FreePBX. disallow and allow section. Trunk Settings, PEER Details Globally in Asterisk SIP Settings module.
          Hide
          mickecarlsson added a comment -

          Codecs can be set prior to Dial command with two variables:

          SIP_CODEC = Set the SIP codec for the inbound (=first) call leg.
          SIP_CODEC_OUTBOUND = Set the SIP codec for the outbound call leg.
          

          Show
          mickecarlsson added a comment - Codecs can be set prior to Dial command with two variables: SIP_CODEC = Set the SIP codec for the inbound (=first) call leg. SIP_CODEC_OUTBOUND = Set the SIP codec for the outbound call leg.
          Hide
          TimmiORG added a comment -

          I vote for it!
          br
          Timmi

          Show
          TimmiORG added a comment - I vote for it! br Timmi
          Hide
          deniq added a comment -

          Take a look at Ticket FREEPBX-5912

          Show
          deniq added a comment - Take a look at Ticket FREEPBX-5912
          Hide
          Martin Anderson added a comment -

          I'm really glad to see some action on this ticket. I look forward to the day when I can do 722 for internal calls and 711 for outside calls!

          Show
          Martin Anderson added a comment - I'm really glad to see some action on this ticket. I look forward to the day when I can do 722 for internal calls and 711 for outside calls!

            People

            • Assignee:
              Andrew Nagy
              Reporter:
              Martin Anderson
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                Updated:

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