[Freeswitch-users] IMT Codec Enforcement

Ahmed habiba ahabiba at gmail.com
Tue Apr 5 17:11:27 MSD 2016


Your kind feedback will be appreciated.


> 
> 
> From: Ahmed habiba <ahabiba at gmail.com>
> Subject: [Freeswitch-users] IMT Codec Enforcement
> Date: April 4, 2016 at 10:37:46 PM GMT+3
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Dears,
> 
> kindly I have the below scenario:
> 
> [SoftPhone1]<——Channel1——>[FreeSwitch1]<——Channel2——>[FreeSwitch2]<——Chanel3——>[SoftPhone2]
> 
> Channel1 codec G722
> 
> Channel3 codec G722
> 
> I want to enforce Channel 3 between FreeSwitch instances to be GSM, I change the Dial plane of freeSwitch1 to and the same for FS2
> 
>                   <action application="set" data=“absolute_codec_string=GSM"/>
> 
>                   <action application="bridge" data="sofia/internal/$1 at 10.8.0.22 <mailto:sofia/internal/$1 at 10.8.0.22>"/>
> 
> However when I run call from FS2 to FS1, FS1 reply with the below putting “a=rtpmap:106 opus/48000/2” as the only available codec as below, your kind help will be appreciated
> 
> 
> 
> recv 1275 bytes from udp/[10.8.0.22]:5060 at 15:22:35.560845:
>    ------------------------------------------------------------------------
>    SIP/2.0 183 Session Progress
>    Via: SIP/2.0/UDP 10.8.0.29;rport=5060;branch=z9hG4bKSr1v2mgcctX5H
>    From: "Extension 1000" <sip:1000 at 10.8.0.29 <sip:1000 at 10.8.0.29>>;tag=U82NjmSa9S9Kg
>    To: <sip:1000 at 10.8.0.22 <sip:1000 at 10.8.0.22>>;tag=g0eNeprr8e4Fg
>    Call-ID: 6d5dbd7f-753d-1234-d78c-bc305bd3840c
>    CSeq: 89562813 INVITE
>    Contact: <sip:1000 at 10.8.0.22:5060;transport=udp <sip:1000 at 10.8.0.22:5060;transport=udp>>
>    User-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 261
>    X-FS-Display-Name: 1000
>    X-FS-Display-Number: sip:1000 at 10.8.0.22 <sip:1000 at 10.8.0.22>
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "1000" <sip:1000 at 10.8.0.22 <sip:1000 at 10.8.0.22>>;party=calling;privacy=off;screen=no
>    
>    v=0
>    o=FreeSWITCH 1459771930 1459771931 IN IP4 10.8.0.22
>    s=FreeSWITCH
>    c=IN IP4 10.8.0.22
>    t=0 0
>    m=audio 25836 RTP/AVP 106 101
>    a=rtpmap:106 opus/48000/2
>    a=fmtp:106 useinbandfec=1; minptime=20
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> Thanks,
> 
> Ahmed Habiba.
> 

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