[asterisk-bugs] [JIRA] (ASTERISK-30269) SRTCP UNPROTECT FAILED (UNHOLD ISSUES)

[Tyler Pearson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30269?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=260511#comment-260511 ] 

Tyler Pearson commented on ASTERISK-30269:
------------------------------------------

the side calling into the PBX can still hear everything. there are no additional steps we are doing to setup the call. 

here is the config for the extension i am testing:

[509]
type=endpoint
aors=509
auth=509-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=g722,ulaw,opus,g729,g726,gsm,alaw
context=from-internal
callerid=Tyler Pearson <509>

dtmf_mode=rfc4733
direct_media=yes
mailboxes=509 at default

mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
max_audio_streams=1
max_video_streams=1
bundle=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
user_eq_phone=yes
send_connected_line=yes
media_encryption=sdes
timers=yes
timers_min_se=500
media_encryption_optimistic=no
refer_blind_progress=yes
refer_blind_progress=yes
rtp_timeout=30
rtp_timeout_hold=300
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en
one_touch_recording=on
record_on_feature=apprecord
record_off_feature=apprecord




> SRTCP UNPROTECT FAILED (UNHOLD ISSUES)
> --------------------------------------
>
>                 Key: ASTERISK-30269
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30269
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_outbound_registration, Resources/res_srtp
>    Affects Versions: 18.15.0
>         Environment: Debian GNU/Linux 10 (buster) in GCP.
>            Reporter: Tyler Pearson
>            Assignee: Tyler Pearson
>
> We just upgraded our phone servers to the new Incredible PBX (Asterisk Version: 18.2.1). I can't find any other form online that has offered a suggestion for a fix to a hold issue we are having.
> When a call comes in, audio on both ends works until that call is placed on hold from our end for what seems to be a certain amount of time; When the call is retrieved from hold, there is no audio on our end, but sometimes the other end can still hear us. I can pretty reliably reproduce the issue by calling from my cell phone into our phone server, placing myself on hold, and then waiting to retrieve that call for about 15-20+ minutes (anytime the call is retrieved before that time, audio resumes okay on both ends). Our phone server at that time then continuously produces the following logs until the call is terminated by either side:
> 19243 [2022-10-04 16:44:21] VERBOSE[21514][C-0000001c] res_srtp.c: SRTCP unprotect failed on SSRC 1667512219 because of replay check failed (index too old)
> 19244 [2022-10-04 16:44:21] VERBOSE[21514][C-0000001c] res_srtp.c: SRTP unprotect failed on SSRC 1667512219 because of authentication failure 160
> This message sometimes also appears:
> 5904 [2022-10-04 11:21:18] WARNING[27806] res_pjsip_outbound_registration.c: No response received from 'sip:**our-telephony-provider.com**:5060' on registration attempt to 'sip:70511898@**our-telephony-provider.com**:5060', retrying in '60'
> Our telephony provider looked a bit into this and said it may be a media session timeout while on hold. They also could not tell from a PCAP of the issue if it was our sever or the softphone we are using (Linphone). They pointed us to contact Asterisk support. I reached out to people on voip-info.org (here is the thread: https://www.voip-info.org/forum/threads/srtcp-unprotect-failed-unhold-issues.26668/)  and they pointed me here.
> Not sure if this is the right place for this, but would someone be able to help me troubleshoot this?



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