[asterisk-bugs] [JIRA] (ASTERISK-26758) res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets

[Kevin Harwell (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26758?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Kevin Harwell updated ASTERISK-26758:
-------------------------------------

    Target Release Version/s: 16.3.0

> res_hep_pjsip: For WebRTC clients Asterisk insert same ip address in "source ip address" and "destination ip address" fields in HEP packets
> -------------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26758
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26758
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_hep, Resources/res_hep_pjsip
>    Affects Versions: 13.13.1, 14.3.0
>         Environment: Centos 7.3 (1611), HOMER 5.0.5, PJPROJECT 2.5.5
>            Reporter: Max Norba
>            Assignee: Joshua C. Colp
>            Severity: Minor
>              Labels: pjsip, webrtc
>      Target Release: 13.19.0, 15.2.0, 16.0.0, 16.3.0
>
>         Attachments: ASTERISK-26758.pcap, ASTERISK-26758.txt
>
>
> For all SIP packets that generate my WebRTC clients Asterisk send in HEP packets same ip address (client ip address) in source and destination field. On call-flow in HOMER it looks like client send all SIP packets to himself. Some example from pcap trace:
> {noformat}
> HEP3 Protocol
> HEP ID: HEP3
> Length (Bytes): 1892
> Protocol family: IPv4
> Protocol ID: UDP
> Source port: 49467
> Destination port: 49467
> Timestamp: 1485262590
> Timestamp us: 903000
> Protocol Type: SIP
> Capture ID: 1234
> Source IP address: 192.168.1.58
> Destination IP address: 192.168.1.58
> Correlation ID: 48607821-99d8-d400-4531-557e671ef1cf
> {noformat}
> For trunk connections, clients that use ip phones or softphones all work as expected. The problem only with clients that use websocket as transport.



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