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

Tue Jun 11 11:10:20 EDT 2019

Thanks for the info. I am using combinations of c,o,r,f,w,i,e

I'll look into it more when I get to it & report back on what works for me
:)

Mark.

On Tue, 11 Jun 2019 at 13:59, Callum Guy <callum.guy at x-on.co.uk> wrote:

> 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>
>>>> >
>>>> > _______________________________________________
>>>> > Users mailing list
>>>> > Users at lists.opensips.org
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>>>> >
>>>>
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>>
>> --
>> Mark Farmer
>> farmorg at gmail.com
>> _______________________________________________
>> Users mailing list
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>>
>
>
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> X-on is a trading name of Storacall Technology Ltd a limited company
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-- 
Mark Farmer
farmorg at gmail.com
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