[asterisk-bugs] [JIRA] (ASTERISK-29936) app_confbridge / translate: unable to build translation path

[Joshua C. Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29936?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp updated ASTERISK-29936:
--------------------------------------

    Assignee: N A  (was: Unassigned)
      Status: Waiting for Feedback  (was: Triage)

And does this occur with Originate in AMI as well, or is this strictly limited to the Originate dialplan application?

> app_confbridge / translate: unable to build translation path
> ------------------------------------------------------------
>
>                 Key: ASTERISK-29936
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29936
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_confbridge, Applications/app_originate, Channels/chan_sip/Video
>    Affects Versions: 18.9.0
>            Reporter: N A
>            Assignee: N A
>         Attachments: translatebug.txt
>
>
> When two peers through a ConfBridge and Originate with the h264 codec try to bridge together, the ConfBridge fails because it's unable to build a translation path. Error is starting codec invalid.
> Before confbridge, the interesting bit is that the write format and read format before are h264, but after the write format is slin/ulaw and the read format is h264. Not sure if that means anything.
> Yes, I know I'm using chan_sip for this, but chan_pjsip fails if I try to set up a video call so that doesn't really pan out.
> Removing h264 from the list of codecs in Originate fixes the problem, but obviously that disables video support.
> With core debug 5, here's what's in that area:
> {noformat}
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_softmix.c:709 softmix_bridge_join:  SIP/ATAxMicroSIP1-00000073:
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: channel.c:5580 ast_set_read_format_path: Channel SIP/ATAxMicroSIP1-00000073 setting read format path: h264 -> slin
> [2022-02-24 20:52:02] WARNING[21379][C-0000009f]: translate.c:494 ast_translator_build_path: No translator path: (starting codec is not valid)
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: channel.c:5826 set_format: Channel SIP/ATAxMicroSIP1-00000073 setting write format path: slin -> ulaw
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: dsp.c:512 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: dsp.c:512 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_channel.c:301 ast_bridge_channel_leave_bridge_nolock: Setting 0x7f01cc523aa0(SIP/ATAxMicroSIP1-00000073) state from:0 to:1
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_softmix.c:772 softmix_bridge_join:
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_softmix.c:2458 softmix_bridge_stream_topology_changed:  SIP/ATAxMicroSIP1-00000073:
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_softmix.c:2466 softmix_bridge_stream_topology_changed:  SIP/ATAxMicroSIP1-00000073: Not in SFU mode
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: stasis_bridges.c:290 bridge_snapshot_update_create: Update: 0x7f01cc300b78  Old: 22bd1255-c2b2-4082-b1c6-44c6203db65b  New: 22bd1255-c2b2-4082-b1c6-44c6203db65b
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: stasis_bridges.c:270 bridge_snapshot_update_dtor: Update: 0x7f01cc300b78  Old: 22bd1255-c2b2-4082-b1c6-44c6203db65b  New: 22bd1255-c2b2-4082-b1c6-44c6203db65b
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: res_rtp_asterisk.c:4398 ast_rtp_change_source: (0x56469edf7d90) RTP changing ssrc from 1365169695 to 1195060383 due to a source change
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: res_rtp_asterisk.c:4402 ast_rtp_change_source: (0x56469edf7d90) RTP changing ssrc for SRTP from 1365169695 to 1195060383
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: res_srtp.c:606 ast_srtp_add_stream: Adding new policy for SSRC 1195060383
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: res_srtp.c:636 ast_srtp_change_source: Couldn't remove stream (13)
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_channel.c:359 ast_bridge_channel_restore_formats: Bridge is returning 0x7f01cc523aa0(SIP/ATAxMicroSIP1-00000073) to write format h264
> [2022-02-24 20:52:02] DEBUG[21383]: channel_internal_api.c:682 ast_channel_nativeformats_set:  <initializing>: Formats: (nothing)
> [2022-02-24 20:52:02] DEBUG[21383]: channel_internal_api.c:692 ast_channel_nativeformats_set:  Channel is being initialized or destroyed
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge.c:972 ast_bridge_destroy: Bridge 22bd1255-c2b2-4082-b1c6-44c6203db65b: telling all channels to leave the party
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge.c:338 bridge_dissolve: Bridge 22bd1255-c2b2-4082-b1c6-44c6203db65b: dissolving bridge with cause 16(Normal Clearing)
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge_channel.c:301 ast_bridge_channel_leave_bridge_nolock: Setting 0x7f01cc252980(CBAnn/oe9-00000213;2) state from:0 to:2
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: bridge.c:299 bridge_queue_action_nodup: Bridge 22bd1255-c2b2-4082-b1c6-44c6203db65b: queueing action type:13 sub:1001
> [2022-02-24 20:52:02] DEBUG[21379][C-0000009f]: pbx.c:2938 pbx_extension_helper: Launching 'DumpChan'
> [2022-02-24 20:52:02]     -- Executing [s at astrex-local:7] DumpChan("SIP/ATAxMicroSIP1-00000073", "") in new stack
> {noformat}



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