Summary:	ASTERISK-25379: no sound on pjsip channel with bridge_native_rtp enabled
Reporter:	Thomas Stein (himbeere)	Labels:
Date Opened:	2015-09-06 08:40:27	Date Closed:	2015-09-06 12:23:37
Priority:	Critical	Regression?
Status:	Closed/Complete	Components:	Channels/chan_pjsip
Versions:	13.5.0	Frequency of Occurrence
Related Issues:
Environment:	Gentoo Linux, Kernel 4.2.0.	Attachments:
Description:	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.
Comments:	By: Asterisk Team (asteriskteam) 2015-09-06 08:40:29.483-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Joshua C. Colp (jcolp) 2015-09-06 10:45:14.552-0500 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 By: Thomas Stein (himbeere) 2015-09-06 12:19:40.965-0500 Hm. I just switched back to a "bridge_native_rtp" enabled installation and suddenly i cannot reproduce the issue. I am a litte bit puzzled now. Lets see if that stays that way. Sorry for the noise so far. I will reopen the bug if the unwanted behaviour returns again. ciao t.