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

Callum Guy callum.guy at x-on.co.uk
Tue Jun 11 08:56:18 EDT 2019


I'm not 100% clear on your situation but if it helps I can confirm that
I've not had to use that option in the past, my configs simply use the same
options as used when rtpproxy was initially offered (in my case
combinations of r,f,i,e - bridging mode) and I've found that it is smart
enough to offer up the same individual proxy and the same ports for a
matching dialog - I presume this is partly the modules responsibility and
partly the rtpproxy software.

Hope that helps

On Tue, 11 Jun 2019 at 12:48, Mark Farmer <farmorg at gmail.com> wrote:

> 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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>>> >
>>>
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>>
>>
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>
> --
> Mark Farmer
> farmorg at gmail.com
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