[asterisk-bugs] [JIRA] (ASTERISK-28416) Unable to get rtp codec payload code for slin
Brian J. Murrell (JIRA)
noreply at issues.asterisk.org
Wed May 15 09:01:48 CDT 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28416?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=247158#comment-247158 ]
Brian J. Murrell commented on ASTERISK-28416:
---------------------------------------------
{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.
> Unable to get rtp codec payload code for slin
> ---------------------------------------------
>
> Key: ASTERISK-28416
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-28416
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Core/RTP
> Affects Versions: 13.26.0
> Reporter: Brian J. Murrell
> Assignee: Brian J. Murrell
> Severity: Trivial
> Labels: fax, pjsip
>
> 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}
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