[asterisk-bugs] [JIRA] (ASTERISK-14934) Asterisk cuts audio to the internal extension

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-14934?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225428#comment-225428 ] 

Joshua Colp commented on ASTERISK-14934:
----------------------------------------

Per the Asterisk versions page [1], the maintenance (bug fix) support for the Asterisk branch you are using has ended. For continued maintenance support please move to a supported branch of Asterisk. After testing with a supported branch, if you find this problem has not been resolved, please open a new issue against the latest version of that Asterisk branch.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions



> Asterisk cuts audio to the internal extension
> ---------------------------------------------
>
>                 Key: ASTERISK-14934
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-14934
>             Project: Asterisk
>          Issue Type: Bug
>          Components: Channels/chan_sip/CodecHandling
>            Reporter: ilker Aktuna
>            Severity: Minor
>         Attachments: agi_fail.txt, all_sip_conf.txt, audio-problem.cap, rtp_debug.txt, trace.txt
>
>
> Hi, 
> I have a problem with one call scenario on my Asterisk server. 
> The call is coming from an external SIP proxy to my server. Asterisk sends the call to one internal extension. 
> Extension rings and then answers. 
> The user at extension hears audio for 0.5 seconds. Then audio in that direction is cut. (no RTP is sent) 
> Audio in other direction continues. 
> I got network traces for this call scenario. Everything seems normal in SIP messaging and as RTP. 
> I can also see that external audio continues even after the internal extension's audio is cut.
> This problem occurs with one type of CPE on the external side. With other CPEs this problem is not reproduced.
> I believe Asterisk cuts the internal extension's audio because it doesn't like something in the RTP stream of external CPE.
> I will add the network trace for unsuccessful call together with RTP debug on Asterisk.
> On both of these files it is clearly seen that one way audio is cut in a very short time after call is established.
> ****** ADDITIONAL INFORMATION ******
> The Asterisk server has 2 interfaces (ppp0 to WAN, and br0 to LAN) 
> So internal clients reach the Asterisk on br0 interface and Asterisk reaches the external proxy through ppp0. 
> In the network trace IP addresses are:
> 192.168.254.254 : Asterisk server LAN address
> 192.168.254.5 : Extension 995 in LAN , Linksys SPA 3000
> 95.65.180.146 : Asterisk server WAN address
> 193.243.202.97 : External SIP Proxy
> 193.243.202.124 : SIP Proxy RTP address



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