[asterisk-bugs] [JIRA] (ASTERISK-25379) no sound on pjsip channel with bridge_native_rtp enabled
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Sun Sep 6 12:21:35 CDT 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-25379?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team updated ASTERISK-25379:
-------------------------------------
Assignee: Asterisk Team (was: Thomas Stein)
Status: Triage (was: Waiting for Feedback)
> 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
> Assignee: Asterisk Team
> 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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