[asterisk-bugs] [JIRA] (ASTERISK-26860) Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)

[Evers Lab (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26860?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=237269#comment-237269 ] 

Evers Lab commented on ASTERISK-26860:
--------------------------------------

{{Asterisk CLI:}}
{noformat}
*CLI>
*CLI>
*CLI>
  == Using SIP RTP CoS mark 5
    -- Executing [999300 at TRUNK-IN:1] Dial("SIP/TRUNK-00000012", "SIP/999300") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/999300
    -- SIP/999300-00000013 is ringing
    -- SIP/999300-00000013 is ringing
[Jun  2 12:31:20] WARNING[5008]: netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
       > 0x7f9a0c005d00 -- Probation passed - setting RTP source address to 10.10.10.22:39728
    -- SIP/999300-00000013 answered SIP/TRUNK-00000012
    -- Channel SIP/999300-00000013 joined 'simple_bridge' basic-bridge <cb282b15-5979-42db-8286-374446dbffcb>
    -- Channel SIP/TRUNK-00000012 joined 'simple_bridge' basic-bridge <cb282b15-5979-42db-8286-374446dbffcb>
       > Bridge cb282b15-5979-42db-8286-374446dbffcb: switching from simple_bridge technology to native_rtp
       > Locally RTP bridged 'SIP/TRUNK-00000012' and 'SIP/999300-00000013' in stack
       > Locally RTP bridged 'SIP/TRUNK-00000012' and 'SIP/999300-00000013' in stack
       > 0x7f9a0c005d00 -- Probation passed - setting RTP source address to 10.10.10.22:39728
       > 0x7f995c00f6d0 -- Probation passed - setting RTP source address to 10.10.10.6:27036
    -- Channel SIP/TRUNK-00000012 left 'native_rtp' basic-bridge <cb282b15-5979-42db-8286-374446dbffcb>
    -- Channel SIP/999300-00000013 left 'native_rtp' basic-bridge <cb282b15-5979-42db-8286-374446dbffcb>
  == Spawn extension (TRUNK-IN, 999300, 1) exited non-zero on 'SIP/TRUNK-00000012'
*CLI>

{noformat}


> Upon RTCP reception, netsock2.c:210 ast_sockaddr_split_hostport: Port missing in (null)
> ---------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26860
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26860
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.14.0
>         Environment: CentOS7
>            Reporter: Evers Lab
>            Assignee: Kevin Harwell
>            Severity: Minor
>      Target Release: 13.16.0, 14.5.0
>
>         Attachments: conf_and_verbose.txt
>
>
> When the call arrives to the Queue and listens to music on hold, the console continually gives an error. When the agent answers the call, the error stops
> [Edit by Rusty - adding Sean Bright's reproduction guide below]
> This is fairly easy to duplicate with Asterisk 13 from git:
> {{extensions.conf:}}
> {noformat}
> [just-moh]
> exten => 1234,1,Answer()
> exten => 1234,n,MusicOnHold()
> {noformat}
> {{Asterisk CLI:}}
> {noformat}
> *CLI> channel originate PJSIP/some-endpoint extension 1234 at just-moh
> {noformat}
> Answer {{some-endpoint}} and place the call on hold. Every time Asterisk receives an RTCP packet, you'll see this error.



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