[asterisk-dev] DirectMedia or canreinvite=UPDATE - asterisk not handling 200K response with SDP for UPDATE sent

[bala murugan]

thanks Matt for the response

i understand the limitation for incoming/inbound UPDATE request .

But this is we asterisk sending UPDATE to bridge the media directly between
caller and callee as an alternate option to RE-INVITE ( canreinvite=update
or directmedia=update)

We are not handling the SDP in 200 OK response for UPDATE we send , it is
just ignored i believe this will be an issue if there is a change in the
media endpoint(port or IP ) , which should not be a limitation , instead i
believe we have to handle the 200 OK response to UPDATE and Update the
connected Peer with SDP change , like we do for RE-INVITE today.

Let me know your comments - i hope i didnt misunderstood .

thanks again
Bala


On Wed, Sep 25, 2013 at 11:57 AM, Matthew Jordan <mjordan at digium.com> wrote:

>
> On Wed, Sep 25, 2013 at 9:57 AM, bala murugan <fightwithme at gmail.com>wrote:
>
>> Hi ,
>>
>>   I am working on asterisk 11.3 and noticed it is not handling the
>> response for UPDATE when trying to do remote bridge using UPDATE . Any
>> change in SDP that comes in 200 OK response is ignored and not update sent
>> to connected peer .
>>
>> Looks like a bug - l looked at the code and i didnt see any code relevant
>> to this scenarion.
>>
>> Can some provide me an input - i am planning to fix this .
>>
>>
> It is not a bug, although it is a limitation of chan_sip.
>
> Asterisk does not advertise support for the UPDATE method. The
> handle_update_request method is quite explicit about this:
>
> /*!
>  * \brief bare-bones support for SIP UPDATE
>  *
>  * XXX This is not even close to being RFC 3311-compliant. We don't
> advertise
>  * that we support the UPDATE method, so no one should ever try sending us
>  * an UPDATE anyway. However, Asterisk can send an UPDATE to change
> connected
>  * line information, so we need to be prepared to handle this. The way we
> distinguish
>  * such an UPDATE is through the X-Asterisk-rpid-update header.
>  *
>  * Actually updating the media session may be some future work.
>  */
>
> Additionally, Asterisk does not list UPDATE in its allowed methods:
>
> #define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
> SUBSCRIBE, NOTIFY, INFO, PUBLISH"
>
> The SIP UA sending an UPDATE request to Asterisk should not be doing so.
>
> If you wanted to add full support for RFC 3311, that would be fine - but
> doing so should be done in trunk, as it is an improvement to chan_sip.
>
> Matt
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
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