[asterisk-dev] storing INVITE fmtp and use it to send relay
Kelvin Chua
kelchy at gmail.com
Mon Jun 29 04:36:16 CDT 2015
Guys,
just tried asterisk13 and added seanbrights' patch for opus.
incoming INVITE has fmtp ------>
maxplaybackrate=8000;sprop-maxcapturerate=8000
but INVITE to my registered peer is
----------> maxplaybackrate=48000;sprop-maxcapturerate=48000
it should not even have to load up the opus patch because it is just a
passthrough.
have you changed anything to chan_sip.c to make this work?
Kelvin Chua
On Sat, Jun 27, 2015 at 7:26 AM, Kelvin Chua <kelchy at gmail.com> wrote:
> i verified parse_sdp is doing its job correctly and stores it in struct.
> but after going back to chan_sip somehow somewhere everything resets before
> generate_sdp. maybe because i am working on ast12, i'm going to try 13
> On Jun 26, 2015 5:49 PM, "Joshua Colp" <jcolp at digium.com> wrote:
>
>> Kelvin Chua wrote:
>>
>>> Just an experiment I am doing, correct me if I am wrong
>>>
>>> If I receive an INVITE with fmtp from a peer, it won't be used to build
>>> the INVITE to the egress right?
>>>
>>> What will happen is, codecs.conf will be checked for the parameters and
>>> use that to build the INVITE.
>>>
>>> Is there any function I can use to get away from this behavior and act
>>> like a proxy and just copy the fmtp from the ingress?
>>>
>>
>> As Alexander mentioned there has to be a specific handler for each codec
>> in order to parse/store/create the specific attributes internally. It's not
>> done for every codec. Asterisk also has to be aware of the codec. This is a
>> bit easier in 13+, but may be possible in earlier versions depending on the
>> amount of storage required.
>>
>> Cheers,
>>
>> --
>> Joshua Colp
>> Digium, Inc. | Senior Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>> --
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