[asterisk-bugs] [JIRA] (ASTERISK-20633) Asterisk SIP channel handling of MOH media re-INVITES not RFC 3264 compliant

[sohosys (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20633?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199855#comment-199855 ] 

sohosys commented on ASTERISK-20633:
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So you maintain that even though a new invite (re-invite) to a different endpoint was sent that there is not a "new session" being established? I understand what RFC3264 is stating, but if you believe that changing one of the endpoints creates a new session it is irrelevant. If this is the belief then it only applies when one of the endpoints in an established media session wish to change the parameters of the established media session. I am not trying to be argumentitive, and I dont have a strong opinion either way, just trying to get everyone to see it the same way so we can all agree on what is right and what is wrong. I guess the definition of "session" as it is used in RFC3264 is the key.
                
> Asterisk SIP channel handling of MOH media re-INVITES not RFC 3264 compliant
> ----------------------------------------------------------------------------
>
>                 Key: ASTERISK-20633
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20633
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 1.8.15.0
>         Environment: asterisk 1.8.15.0 on CentOS 5
>            Reporter: sohosys
>            Assignee: Michael L. Young
>            Severity: Minor
>         Attachments: mohdebug.pdf
>
>
> When asterisk is connected to a SIP peer that uses a re-INVITE to place a call on Music On Hold, but omits the "a=sendrecv" tag in the SDP the handling of the SDP offer by Asterisk is not RFC 3264 compliant. This is the case when interoperating with Broadsoft Broadworks , Cisco Call Manager and presumably some other IP PBXs.
> When the call is placed on hold the connected Broadworks platform first requests that the current media is stopped by sending a re-INVITE with a=”inactive” in the SDP. Asterisk accepts this SDP offer and starts MOH. The connected Broadworks platform then sends another re-INVITE with a new media endpoint IP address and port belonging to a media server that is streaming the MOH (this re-INVITE does not contain “a=sendrecv” in the SDP, but does contain a new media endpoint IP and port). Asterisk ACKs the SDP offer with “a=inactive” based on what appears to be logic that says “the media was previously inactive, and the re-INVITE does not explicitly request sendrecv, so we assume that the intention is to leave the media inactive”.
> According to RFC 3264 “If the offerer wishes to both send and receive media with its peer, it MAY include an a=sendrecv" attribute, or it MAY omit it, since sendrecv is the default.”
> The RFC compliant method of handling this SDP offer would be to evaluate each re-INVITE without regard to the previous media state and assume the default of sendrecv when there is no a=inactive, a=sendonly, or a=recvonly in the SDP.
> I have confirmed with both Broadsoft and Cisco that their products by design do not send a=sendrecv in the SDP when directing media to a media server to playback MOH or IVR prompts, and they both claim that this is RFC compliant, and the RFC supports their claims.

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