Summary: | ASTERISK-28416: Unable to get rtp codec payload code for slin | ||||
Reporter: | Brian J. Murrell (brian_j_murrell) | Labels: | fax pjsip | ||
Date Opened: | 2019-05-14 06:33:13 | Date Closed: | 2020-09-15 14:12:29 | ||
Priority: | Trivial | Regression? | |||
Status: | Closed/Complete | Components: | Core/RTP | ||
Versions: | 13.26.0 | Frequency of Occurrence | Constant | ||
Related Issues: |
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Environment: | Attachments: | ||||
Description: | When making a call to a certain endpoint asterisk logs:
{noformat} [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for slin [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for testlaw [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk [May 14 07:26:28] WARNING[12084]: res_pjsip_sdp_rtp.c:1296 create_outgoing_sdp_stream: Unable to get rtp codec payload code for silk {noformat} | ||||
Comments: | By: Asterisk Team (asteriskteam) 2019-05-14 06:33:13.893-0500 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: George Joseph (gjoseph) 2019-05-15 08:23:33.521-0500 "silk" by itself isn't a valid format. It needs to be "silk8", "silk12", etc. "slin" is valid though. Please provide the full configuration for the endpoint. You can just paste the results of {{pjsip show endpoint <endpoint>}}. What type of device is the actual endpoint? Does the call succeed with good audio despite the warnings? By: Brian J. Murrell (brian_j_murrell) 2019-05-15 09:00:14.177-0500 {noformat} Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.> I/OAuth: <AuthId/UserName...........................................................> Aor: <Aor............................................> <MaxContact> Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..> Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................> Identify: <Identify/Endpoint.........................................................> Match: <criteria.........................> Channel: <ChannelId......................................> <State.....> <Time.....> Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......> ========================================================================================== Endpoint: outgoing Unavailable 0 of inf Aor: outgoing 0 ParameterName : ParameterValue =========================================================================================================================================================================================================================================================================== 100rel : yes accept_multiple_sdp_answers : false accountcode : acl : aggregate_mwi : true allow : (g723|ulaw|alaw|gsm|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|h263|h263p|h264|mpeg4|vp8|vp9|red|t140|silk|silk|silk|silk) allow_overlap : true allow_subscribe : true allow_transfer : true aors : outgoing asymmetric_rtp_codec : false auth : bind_rtp_to_media_address : false call_group : callerid : <unknown> callerid_privacy : allowed_not_screened callerid_tag : connected_line_method : invite contact_acl : context : default 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 : false 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 : none fax_detect : false fax_detect_timeout : 0 follow_early_media_fork : true force_avp : false force_rport : true from_domain : from_user : g726_non_standard : false ice_support : true identify_by : username,ip ignore_183_without_sdp : false inband_progress : false incoming_mwi_mailbox : 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 : mwi_subscribe_replaces_unsolicited : no named_call_group : named_pickup_group : notify_early_inuse_ringing : false one_touch_recording : false outbound_auth : outbound_proxy : pickup_group : record_off_feature : automixmon record_on_feature : automixmon refer_blind_progress : true rewrite_contact : true rpid_immediate : false rtcp_mux : false rtp_engine : asterisk rtp_ipv6 : false rtp_keepalive : 0 rtp_symmetric : true rtp_timeout : 0 rtp_timeout_hold : 0 sdp_owner : - sdp_session : Asterisk send_connected_line : yes send_diversion : true send_pai : false send_rpid : false set_var : srtp_tag_32 : false sub_min_expiry : 0 subscribe_context : suppress_q850_reason_headers : false t38_udptl : true 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 : trust_connected_line : yes trust_id_inbound : false trust_id_outbound : false use_avpf : false use_ptime : false user_eq_phone : false voicemail_extension : {noformat} The {{allow}} list is not lost on me. I'm just not sure why it is that way, with so many (and repeated) codecs. {quote} What type of device is the actual endpoint? {quote} In the case of the call that displays these warnings, it's linphone-android. {quote} Does the call succeed with good audio despite the warnings {quote} Indeed it does succeed and the audio is good. By: George Joseph (gjoseph) 2019-05-15 10:33:51.155-0500 OK. I'm guessing you have "all" specified for "allow" and for some reason we're truncating the sample rate off the end of the formats so "slin", "slin12", "slin16", "slin24", etc all show as just "slin". That is indeed a bug. You may want to not use "all" and just stick to specific codecs if for no other reason than to prevent Astrerisk from having to try and process all of them during call negotiation. By: Friendly Automation (friendly-automation) 2020-09-15 14:12:31.269-0500 Change 14922 merged by Friendly Automation: format_cap: Perform codec lookups by pointer instead of name [https://gerrit.asterisk.org/c/asterisk/+/14922|https://gerrit.asterisk.org/c/asterisk/+/14922] By: Friendly Automation (friendly-automation) 2020-09-15 14:36:38.565-0500 Change 14920 merged by George Joseph: format_cap: Perform codec lookups by pointer instead of name [https://gerrit.asterisk.org/c/asterisk/+/14920|https://gerrit.asterisk.org/c/asterisk/+/14920] By: Friendly Automation (friendly-automation) 2020-09-15 14:38:12.113-0500 Change 14941 merged by George Joseph: format_cap: Perform codec lookups by pointer instead of name [https://gerrit.asterisk.org/c/asterisk/+/14941|https://gerrit.asterisk.org/c/asterisk/+/14941] By: Friendly Automation (friendly-automation) 2020-09-15 14:38:49.216-0500 Change 14942 merged by George Joseph: format_cap: Perform codec lookups by pointer instead of name [https://gerrit.asterisk.org/c/asterisk/+/14942|https://gerrit.asterisk.org/c/asterisk/+/14942] By: Friendly Automation (friendly-automation) 2020-09-15 14:39:02.818-0500 Change 14943 merged by George Joseph: format_cap: Perform codec lookups by pointer instead of name [https://gerrit.asterisk.org/c/asterisk/+/14943|https://gerrit.asterisk.org/c/asterisk/+/14943] |