[Freeswitch-users] How to configure G722 for internal and G729 for external calls

[Jason Lewis]

Hi Bote,

Appologies for the long log output. I'll use the pastebin in future.

I have purchased and installed 2 G729 licenses. I have tested and shown
that I can use them in my configuration.

Hi Bote,

Thanks for your guidance, and sorry for the long log post.

I eventually got it working by setting the absolute_code_string in the
dialplan.

    <action application="export" data="absolute_codec_string=G729"/>
 
Thanks to BKW and Drestreyf for help on that.

Jason



Bote Man wrote on 10/03/2015 4:46 PM:
> 1) Please don't post extensive log output to the mailing list. The developers much prefer that you use the FreeSWITCH pastebin and choose the FreeSWITCH log syntax highlighting:
> https://pastebin.freeswitch.org/
> and pay close attention to the instructions in the prompt for credentials that pops up.
>
> 2) I see:
> 2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec string G729 at 8000h@20i not supported on sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance
>
> Did you license G.729 codec? Because FS will need to transcode from the G.722 used internally to G.729 towards your carrier. My guess is that this is where your problem lies.
>
> With the vanilla configuration I believe that FS can agree on G.729 with the far end as long as it is merely passing the RTP stream through the switch untouched. Once FS needs to transcode it between G.729 and G.722 you need to fork over money for the number of simultaneous G.729 calls that you expect since it is a commercially restricted codec.
>
> Details on the Confluence wiki at:
> https://freeswitch.org/confluence/display/FREESWITCH/mod_com_g729
>
> Of course, I could be totally wrong about this, but if you are down under then you'll be asleep when the everybody else wakes up so I figure I'd give it a stab to give you a head-start.
>
> Bote
>
>  
>
>
> -----Original Message-----
> From: Jason Lewis
> Sent: Monday, 09 March, 2015 23:19
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] How to configure G722 for internal and G729 for external calls
>
> hi,
>
> I'm trying to get freeswitch to use G722 for internal calls and G729 for
> external calls.
>
> I'm using vanilla.
>
> Am I missing something obvious here?
> .
> .
> .
> 2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4697 [zrtp_passthru] Setting a-leg inherit_codec=true
> 2015-03-10 12:37:37.775786 [DEBUG] mod_sofia.c:4700 [zrtp_passthru] Setting b-leg absolute_codec_string='G722 at 8000h@20i at 64000b,PCMU at 8000h@20i at 64000b,PCMA at 8000h@20i at 64000b'
> ...
> 2015-03-10 12:37:37.775786 [DEBUG] sofia_glue.c:1232 sofia/external/775 sending invite version: 1.4.15 -1 64bit
> Local SDP:
> v=0
> o=FreeSWITCH 1425927473 1425927474 IN IP4 aa.bbb.ccc.ddd
> s=FreeSWITCH
> c=IN IP4 aa.bbb.ccc.ddd
> t=0 0
> m=audio 23984 RTP/AVP 18 101 13
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> 2015-03-10 12:37:38.735794 [DEBUG] sofia.c:6624 Remote SDP:
> v=0
> o=Sippy 2433467651418324771 2 IN IP4 202.85.243.105
> s=session
> t=0 0
> m=audio 10948 RTP/AVP 18 101
> c=IN IP4 202.85.243.53
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
>
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G729:18:8000:20:8000:1]/[G729:18:8000:20:8000:1]
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G729:18:8000:20:8000:1] ++++ is saved as a match
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:2473 Set Codec sofia/external/775 G729/8000 20 ms 160 samples 8000 bits 1 channels
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_codec.c:111 sofia/external/775 Original read codec set to G729:18
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/external/775] 10.0.2.145 port 23984 -> 202.85.243.53 port 10948 codec: 18 ms: 20
> 2015-03-10 12:37:38.735794 [DEBUG] switch_rtp.c:3548 Starting timer [soft] 160 bytes per 20ms
> ...
> 2015-03-10 12:37:38.735794 [NOTICE] sofia.c:7475 Channel [sofia/external/775] has been answered
> 2015-03-10 12:37:38.735794 [DEBUG] switch_channel.c:3689 (sofia/external/775) Callstate Change DOWN -> ACTIVE
> 2015-03-10 12:37:38.735794 [DEBUG] switch_ivr_originate.c:415 Codec string G729 at 8000h@20i not supported on sofia/internal/1001 at freeswitch.xyz.com.au, skipping inheritance
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3627 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
> 2015-03-10 12:37:38.735794 [DEBUG] switch_core_media.c:3682 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
> 2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:2473 Set Codec sofia/internal/1001 at freeswitch.xyz.com.au G722/8000 20 ms 160 samples 64000 bits 1 channels
> 2015-03-10 12:37:38.755785 [DEBUG] switch_core_codec.c:111 sofia/internal/1001 at freeswitch.xyz.com.au Original read codec set to G722:9
> 2015-03-10 12:37:38.755785 [DEBUG] switch_core_media.c:5141 AUDIO RTP [sofia/internal/1001 at freeswitch.xyz.com.au] 10.0.2.145 port 20762 -> 10.0.2.129 port 11794 codec: 9 ms: 20
> 2015-03-10 12:37:38.755785 [DEBUG] switch_channel.c:3399 (sofia/internal/1001 at freeswitch.xyz.com.au) Callstate Change RINGING -> EARLY
> 2015-03-10 12:37:38.755785 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/1001 at freeswitch.xyz.com.au:
> v=0
> o=FreeSWITCH 1425930696 1425930697 IN IP4 10.0.2.145
> s=FreeSWITCH
> c=IN IP4 10.0.2.145
> t=0 0
> m=audio 20762 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
>
>

-- 
Jason Lewis
http://emacstragic.net


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