[Freeswitch-users] IMT Codec Enforcement

[Ahmed habiba]

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"/>

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>;tag=U82NjmSa9S9Kg
   To: <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>
   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
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "1000" <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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