[OpenSIPS-Users] re-INVITE (Hold) looses audio

[Mark Farmer]

Thank you both for your replies.
I realised eventually that I was indeed missing rtpproxy_offer/answer for
the RE-INVITEs.

Callum, I am using rtpproxy_offer/answer but I will keep that in mind for
the future.

Seems to be working now although I think I need to tune the options in this
case. Is there a generally recognised method of re-engaging rtpproxy when
sessions already exist? I'm thinking the 'l' flag is required?

Regards
Mark.


On Tue, 11 Jun 2019 at 09:16, Callum Guy <callum.guy at x-on.co.uk> wrote:

> Hi Mark,
>
> Can you confirm which engagement functions you are using? Does the SDP
> look like its being handled for the RE-INVITE transactions?
>
> I have dealt with some similar scenarios recently and would highly
> recommend using the separate rtpproxy_offer() and rtpproxy_answer() methods
> over the dialog bound rtpproxy_engage() - this will give you a more
> granular control. Generally speaking I would advise that you identity a
> loose routed INVITE and check for present of SDP and use this to trigger an
> offer, somewhat like this:
>
> if (loose_route() && is_method("INVITE") && has_body("application/sdp")) {
>   rtpproxy_offer(OPTS);
> }
>
> You'll then need to consider the additional scenarios for last SDP and ACK
> processing to handle both the answer and edge cases however hopefully this
> will help you move forward in your config.
>
> Good luck,
>
> Callum
>
>
> On Mon, 10 Jun 2019 at 19:55, Răzvan Crainea <razvan at opensips.org> wrote:
>
>> Hi, Mark!
>>
>> Are you engaging RTPProxy for the re-INVITEs too? Can you check if
>> RTPProxy times out the session due to lack of audio from one side? You
>> can see that in the logs.
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> On 6/10/19 4:23 PM, Mark Farmer wrote:
>> > Hi all
>> >
>> > I'm trying to solve an issue where if the call is placed on hold via a
>> > re-INVITE, my end stops the audio at that point so when another
>> > re-INVITE is received to resume the call, the audio does not restart. I
>> > can still hear the other party but they cannot hear me because I'm not
>> > sending any RTP any more.
>> >
>> > I'm really struggling to pin this one down. Can anyone give me any
>> clues
>> > as to why this might be happening?
>> >
>> > I'm using OpenSIPS 2.4.5 with RTPProxy 2.0
>> >
>> > Many thanks
>> > Mark.
>> >
>> >
>> > --
>> > Mark Farmer
>> > farmorg at gmail.com <mailto:farmorg at gmail.com>
>> >
>> > _______________________________________________
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>> > Users at lists.opensips.org
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>> >
>>
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-- 
Mark Farmer
farmorg at gmail.com
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