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. |