[asterisk-bugs] [JIRA] (ASTERISK-25379) no sound on pjsip channel with bridge_native_rtp enabled

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25379?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=227499#comment-227499 ] 

Joshua Colp commented on ASTERISK-25379:
----------------------------------------

Thanks for the report and debug. However we also need protocol specific debug captured at the time of the issue. Please include the following:

* Asterisk log files generated using the instructions on the Asterisk wiki [1], with the appropriate protocol debug options enabled, e.g. 'pjsip set logger on' if the issue involves the chan_pjsip channel driver.
* Configuration information for the relevant channel driver, e.g. pjsip.conf.
* A wireshark compatible packet capture, captured at the same time as the Asterisk log output.

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> no sound on pjsip channel with bridge_native_rtp enabled
> --------------------------------------------------------
>
>                 Key: ASTERISK-25379
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25379
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.5.0
>         Environment: Gentoo Linux, Kernel 4.2.0.
>            Reporter: Thomas Stein
>            Severity: Critical
>
> Hello.
> I'm getting no sound out of my phone (pjsip endoint) while bridge_native_rtp is enabled. As soon as i disable the function in menuconfig everything works as expected. 
> Here is my enpoint info:
> ParameterName                 : ParameterValue
>  ====================================================
>  100rel                        : yes
>  accountcode                   : 
>  aggregate_mwi                 : true
>  allow                         : (ulaw|g722|gsm)
>  allow_subscribe               : true
>  allow_transfer                : true
>  aors                          : 501
>  auth                          : auth501
>  call_group                    : 
>  callerid                      : <unknown>
>  callerid_privacy              : allowed_not_screened
>  callerid_tag                  : 
>  connected_line_method         : invite
>  context                       : local
>  cos_audio                     : 0
>  cos_video                     : 0
>  device_state_busy_at          : 0
>  direct_media                  : false
>  direct_media_glare_mitigation : none
>  direct_media_method           : invite
>  disable_direct_media_on_nat   : true
>  dtls_ca_file                  : 
>  dtls_ca_path                  : 
>  dtls_cert_file                : 
>  dtls_cipher                   : 
>  dtls_fingerprint              : SHA-256
>  dtls_private_key              : 
>  dtls_rekey                    : 0
>  dtls_setup                    : active
>  dtls_verify                   : No
>  dtmf_mode                     : rfc4733
>  fax_detect                    : false
>  force_avp                     : false
>  force_rport                   : true
>  from_domain                   : 
>  from_user                     : 
>  g726_non_standard             : false
>  ice_support                   : false
>  identify_by                   : username
>  inband_progress               : false
>  language                      : en
>  mailboxes                     : 
>  media_address                 : 
>  media_encryption              : no
>  media_encryption_optimistic   : false
>  media_use_received_transport  : false
>  message_context               : 
>  moh_suggest                   : default
>  mwi_from_user                 : 
>  named_call_group              : 
>  named_pickup_group            : 
>  one_touch_recording           : false
>  outbound_auth                 : 
>  outbound_proxy                : 
>  pickup_group                  : 
>  record_off_feature            : automixmon
>  record_on_feature             : automixmon
>  rewrite_contact               : true
>  rpid_immediate                : false
>  rtp_engine                    : asterisk
>  rtp_ipv6                      : false
>  rtp_keepalive                 : 3
>  rtp_symmetric                 : true
>  rtp_timeout                   : 0
>  rtp_timeout_hold              : 0
>  sdp_owner                     : -
>  sdp_session                   : Asterisk
>  send_diversion                : true
>  send_pai                      : false
>  send_rpid                     : false
>  set_var                       : 
>  srtp_tag_32                   : false
>  sub_min_expiry                : 0
>  t38_udptl                     : false
>  t38_udptl_ec                  : none
>  t38_udptl_ipv6                : false
>  t38_udptl_maxdatagram         : 0
>  t38_udptl_nat                 : false
>  timers                        : yes
>  timers_min_se                 : 90
>  timers_sess_expires           : 1800
>  tone_zone                     : 
>  tos_audio                     : 0
>  tos_video                     : 0
>  transport                     : transport-udp
>  trust_id_inbound              : true
>  trust_id_outbound             : false
>  use_avpf                      : false
>  use_ptime                     : false
>  user_eq_phone                 : false
> I can provide more information if needed.
> thanks and cheers
> t.



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