[asterisk-bugs] [JIRA] (ASTERISK-28420) In WebRTC video call scenario, packet loss lead to frozen video。

[Benjamin Keith Ford (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28420?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Benjamin Keith Ford updated ASTERISK-28420:
-------------------------------------------

    Assignee: Aaron An
      Status: Waiting for Feedback  (was: Triage)

What are you referring to with "network is pool"? What version of libsrtp are you running? Are you using Cyber Mega Phone? Is it a standard video call? How many endpoints are in the call?

> In WebRTC video call scenario, packet loss lead to frozen  video。
> -----------------------------------------------------------------
>
>                 Key: ASTERISK-28420
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28420
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_srtp
>    Affects Versions: 16.3.0
>         Environment: CentOS 7.5
>            Reporter: Aaron An
>            Assignee: Aaron An
>              Labels: webrtc
>
> In WebRTC video call scenario, calls from Chrome to asterisk. When network is pool, asterisk reports warnings like "SRTP unprotect failed" "SRTP try to re-create" and then the video is frozen, the same time asterisk console report "SRTCP unprotect failed on SSRC xxx" every 1-2 seconds until the call ended. I have investigated this issue for several days and find that there is something wrong with the srtp re-create process. The srtp->policy is store in the hash buckets which is initialized with 5.  This should change from 5 to 1 to avoid indeterminacy policy order when re-create the srtp session。
> in res_srtp.c function res_srtp_new()
> srtp->policies = ao2_t_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, 5,
> 		policy_hash_fn, NULL, policy_cmp_fn, "SRTP policy container");



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