media_mix_inbound_outbound_code=true

https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation#CodecNegotiation-MixingMedia/Codecsondifferentlegs(transcoding)

On May 31, 2016, at 9:41 AM, Kim Rostgaard Christensen <krc@retrospekt.dk> wrote:

We've had a sort-a breakthrough:

If we override the codec string
originate loopback/5900/default &bridge({absolute_codec_string='PCMA,PCMU'}sofia/gateway/gw/number)

We can get the gateway to accept our call. This is most like due to them not supporting L16 - which makes the BEARERCAPABILITY_NOTAVAIL make sense. Yay!

SDP then becomes:

v=0
o=FreeSWITCH 1464414375 1464414376 IN IP4 192.168.1.4
s=FreeSWITCH
c=IN IP4 192.168.1.4
t=0 0
m=audio 24744 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

And everything is fine. Sound comes through, and connection is stable.

This raises two questions:

1. We would really like the default behaviour to use PCMA/8000 for outbound bridged channels.
How do we go about this?

2. Is this intended that a minor release should have such large impact on the SDP and, if so, can anyone elaborate on this?

Best
Kim


On 2016-05-27 16:32, Kim Rostgaard Christensen wrote:
We finally got around to getting a SIP trace.

The only difference we can find is in the SDP

1.6.2:

v=0
o=FreeSWITCH 1464328347 1464328348 IN IP4 192.168.1.3
s=FreeSWITCH
c=IN IP4 192.168.1.3
t=0 0
m=audio 28898 RTP/AVP 102 103 9 0 8 3 101 13 104 105
a=rtpmap:102 L16/8000
a=rtpmap:103 opus/48000/2
a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=20; maxptime=20; samplerate=48000; sprop-stereo=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:104 telephone-event/48000
a=fmtp:104 0-16
a=rtpmap:105 CN/48000
a=ptime:20

1.6.8:

v=0
o=FreeSWITCH 1464330840 1464330841 IN IP4 192.168.1.4
s=FreeSWITCH
c=IN IP4 192.168.1.4
t=0 0
m=audio 28118 RTP/AVP 102 101 13
a=rtpmap:102 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20


Any hints?


On 2016-02-10 18:16, Thomas Løcke wrote:
Hi Ken,

It doesn't work from our dialplan either:

<extension name="foobar" continue="true">
      <condition field="destination_number" expression="^external_transfer_(\d+)$">
        <action application="set" data="ringback=${dk-ring}"/>
        <action application="ring_ready" />
        <action application="bridge" data="[fifo_music=default]sofia/gateway/${default_trunk}/$1"/>
        <action application="hangup"/>
      </condition>
    </extension>

That works swimmingly with 1.6.2, but fails with BEARERCAPABILITY_NOTAVAIL with 1.6.6.


2016-02-10 18:07 GMT+01:00 Ken Rice <krice@freeswitch.org>:
I doubt that was ever an intended feature... why not just send the A leg out and park the bleg just reverse your notation there

On Wed, Feb 10, 2016 at 10:48 AM, Thomas Løcke <thomas.granvej6@gmail.com> wrote:
Hi all,

Using 1.6.2 this works from fs_cli and dialplan:

originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx)

Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL.


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_________________________________________________________________________
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