[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

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26758?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=234965#comment-234965 ] 

Matt Jordan commented on ASTERISK-26758:
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I'll echo what [~rnewton] asked for - I'll need a pcap of the SIP message traffic. A full pjsip log with debug information would also be helpful.

That being said, there's a **very** good chance that there won't be much I can do. With WebRTC, you can't always trust the IP information that the client presents. We'll have to see if there's anything in the SIP message traffic or in the PJSIP transport layer that is reliable.

> 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
>         Environment: Centos 7.3 (1611), HOMER 5.0.5, PJPROJECT 2.5.5
>            Reporter: Max Norba
>            Assignee: Matt Jordan
>            Severity: Minor
>
> 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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