[asterisk-bugs] [JIRA] (ASTERISK-28551) IPv4 address in SDP o= is (null) when configured for NAT using pjsip
Brian J. Murrell (JIRA)
noreply at issues.asterisk.org
Mon Sep 23 19:19:47 CDT 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-28551?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248110#comment-248110 ]
Brian J. Murrell commented on ASTERISK-28551:
---------------------------------------------
{noformat}
server*CLI> pjsip show endpoint brian_h8
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: brian_h8 Not in use 0 of inf
InAuth: brian_h8/brian_h8
Aor: brian_h8 1
Contact: brian_h8/sip:brian_h8 at 10.75.22.32:57886;tr 3a9cc4d1c3 Unknown nan
ParameterName : ParameterValue
=========================================================
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|gsm)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
aors : brian_h8
asymmetric_rtp_codec : false
auth : brian_h8
bind_rtp_to_media_address : false
call_group :
callerid : <unknown>
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : internal-sip
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : true
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 : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain : pbx.example.com
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 :
mailboxes : 2000
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context : messages
moh_suggest : default
mwi_from_user : 2003
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 : false
rpid_immediate : false
rtcp_mux : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : false
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 : 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 :
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : true
use_ptime : false
user_eq_phone : false
voicemail_extension :
{noformat}
It's not lost on me that {{transport:}} above is empty. But that's because I don't want to assign a specific transport but allow the client to decide what to use.
I did try assigning the specific transport from the *Description* to the endpoint in question with:
{{transport=transport-tcp}}
but that didn't seem to resolve the issue.
No {{ice_host_candidates}} defined in {{rtp.conf}}.
> IPv4 address in SDP o= is (null) when configured for NAT using pjsip
> --------------------------------------------------------------------
>
> Key: ASTERISK-28551
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-28551
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: pjproject/pjsip
> Affects Versions: 13.28.1
> Reporter: Brian J. Murrell
> Assignee: Brian J. Murrell
> Labels: fax, pjsip
>
> If I configure my pjsip transport to handle NAT from the Internet:
> {noformat}
> [transport-tcp]
> type=transport
> protocol=tcp
> bind=10.75.22.8:5060
> local_net=10.75.22.0/24
> external_media_address=[external address redacted]
> external_signaling_address=[external address redacted]
> {noformat}
> When a call comes from a TCP registered SIP client on the Internet,
> Asterisk is setting the IPv4 address in the {{o=}} and {{c=}} lines of the
> SDP ICE payload to {{(null)}}:
> {noformat}
> v=0
> o=- 3654 548 IN IP4 (null)
> s=Asterisk
> c=IN IP4 (null)
> {noformat}
> IPv4 addresses in all of the {{a=}} lines are still correct.
> This {{(null)}} of course causes the caller to fail to complete the call.
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