[asterisk-bugs] [JIRA] (ASTERISK-30269) res_srtp: One way audio after unhold

[Tyler Pearson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30269?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=260620#comment-260620 ] 

Tyler Pearson edited comment on ASTERISK-30269 at 11/10/22 3:58 PM:
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EUREKA! IT WORKS!! looks like Joshua and Micheal's suggestion (https://github.com/asterisk/asterisk/blob/master/configs/samples/rtp.conf.sample#L48) was the solution! I had do do a few things different to get it to work with our server: I had to edit /etc/asterisk/rtp_custom.conf and add the line "srtpreplayprotection=no" and then restarting asterisk with init.d. 

The only oddity with this fix is that after 15 minutes on the call, the error still shows in the logs and the audio takes about 2ish seconds to re-establish the INVITES and then the errors stop. Everything works like a charm after that. 

I believe it is just a bug with Linphone. I will see if i can reach out to them to report this. Thank you guys so much for taking the time to help me sort through this issue!


was (Author: tyler_pearson):
EUREKA! IT WORKS!! looks like Joshua's suggestion (https://github.com/asterisk/asterisk/blob/master/configs/samples/rtp.conf.sample#L48) was the solution! I had do do a few things different to get it to work with our server: I had to edit /etc/asterisk/rtp_custom.conf and add the line "srtpreplayprotection=no" and then restarting asterisk with init.d. 

The only oddity with this fix is that after 15 minutes on the call, the error still shows in the logs and the audio takes about 2ish seconds to re-establish the INVITES and then the errors stop. Everything works like a charm after that. 

I believe it is just a bug with Linphone. I will see if i can reach out to them to report this. Thank you guys so much for taking the time to help me sort through this issue!

> res_srtp: One way audio after unhold
> ------------------------------------
>
>                 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: Unassigned
>         Attachments: Screenshot_linphone1.png, Screenshot_linphone2.png
>
>
> 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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