[Freeswitch-users] how to transcode from g.729 to g.711a ?

Mon Jun 22 13:41:29 MSD 2015

G.729 support is not coming out of the box, because you need to
purchase a G.729 license for your FreeSWITCH box.

Without the license, the server can only transparently pass the media.

Maybe there are other codecs that your phone supports, like iLBC or
G722? They don't require the licensing.

On Mon, Jun 22, 2015 at 10:17 AM, Jeff Chua <jeff.chua.linux at gmail.com> wrote:
> I'm attempting to call from a SIP phone (g.729) to the SIP gateway
> that connects g.711a to the PSTN. It's resulting in "SERVICE
> UNAVAILABLE" ....
>
> If I switch my SIP phone to use g.711a, everything works fine.
>
> Any sample on how to transcode 9.729 from my SIP phone to the
> freeswitch server that would then transcode from g.729 to g.711a to
> connect to the SIP gateway?
>
> I've tried g.729 from my SIP phone to call 1000 and 5000 on the
> freeswitch server and that worked. And I've bought license for g.729.
>
> Thanks,
> Jeff.
>
>    ------------------------------------------------------------------------
>    INVITE sip:+6568465283 at 155.161.200.188 SIP/2.0
>    Via: SIP/2.0/TCP
> 155.161.11.194:54927;rport;branch=z9hG4bKPjDNkdtN2t3.xjdgQ-78gpmFRuBhnjA9Qd;alias
>    Max-Forwards: 70
>    From: "1001" <sip:1001 at 155.161.200.188>;tag=x0m7o1ogjzCoro6dBITkX2dB5U.4DX-y
>    To: <sip:+6568465283 at 155.161.200.188>
>    Contact: <sip:1001 at 155.161.11.194:54927;transport=TCP;ob>
>    Call-ID: kQZCHnkGqshj5lmFItexEB8zNG1MA4W6
>    CSeq: 21273 INVITE
>    Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
> NOTIFY, REFER, MESSAGE, OPTIONS
>    Supported: replaces, 100rel, norefersub
>    User-Agent: Bria iOS 3.3.4
>    Content-Type: application/sdp
>    Content-Length:   234
>
>    v=0
>    o=- 3643948404 3643948404 IN IP4 155.161.11.194
>    s=cpc_med
>    c=IN IP4 155.161.11.194
>    t=0 0
>    m=audio 4004 RTP/AVP 18 101
>    a=sendrecv
>    a=rtpmap:18 G729/8000
>    a=fmtp:18 annexb=no
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    ------------------------------------------------------------------------
>    INVITE sip:68465283 at 155.161.200.188:5070 SIP/2.0
>    Via: SIP/2.0/UDP 155.161.200.188:5080;rport;branch=z9hG4bKKX98pj6UB2crp
>    Max-Forwards: 69
>    From: "Extension 1001"
> <sip:FreeSWITCH at 155.161.200.188:5070>;tag=KKS78e6eNpUpN
>    To: <sip:68465283 at 155.161.200.188:5070>
>    Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33
>    CSeq: 77143746 INVITE
>    Contact: <sip:gw+outbound1 at 155.161.200.188:5080;transport=udp;gw=outbound1>
>    User-Agent: FreeSWITCH-mod_sofia/1.5.15b~64bit
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
> REGISTER, REFER, NOTIFY
>    Supported: timer, path, replaces
>    Allow-Events: talk, hold, conference, refer
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 231
>    X-FS-Support: update_display,send_info
>    Remote-Party-ID: "Extension 1001"
> <sip:1001 at 155.161.200.188:5070>;party=calling;screen=yes;privacy=off
>
>    v=0
>    o=FreeSWITCH 1434929593 1434929594 IN IP4 155.161.200.188
>    s=FreeSWITCH
>    c=IN IP4 155.161.200.188
>    t=0 0
>    m=audio 30028 RTP/AVP 18 101 13
>    a=rtpmap:18 G729/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP
> 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp
>    From: "Extension 1001"
> <sip:FreeSWITCH at 155.161.200.188:5070>;tag=KKS78e6eNpUpN
>    To: <sip:68465283 at 155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324
>    Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33
>    CSeq: 77143746 INVITE
>    Contact: <sip:68465283 at 155.161.200.188:5070>
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>    SIP/2.0 503 Service Unavailable
>    Via: SIP/2.0/UDP
> 155.161.200.188:5080;received=155.161.200.188;rport=5080;branch=z9hG4bKKX98pj6UB2crp
>    From: "Extension 1001"
> <sip:FreeSWITCH at 155.161.200.188:5070>;tag=KKS78e6eNpUpN
>    To: <sip:68465283 at 155.161.200.188:5070>;tag=sipcontrol_2457501727-2035324
>    Call-ID: a3a821ca-9356-1233-e59d-0018d2d04e33
>    CSeq: 77143746 INVITE
>    Contact: <sip:68465283 at 155.161.200.188:5070>
>    Retry-After: 1
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
>
>
> My config ...
>
> # vars.xml
>   <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMA"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>
>
> # dialplan/default.xml
>     <extension name="oubout_pstn">
>         <condition field="destination_number" expression="^\+65(.+)$">
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="bridge" data="sofia/gateway/outbound1/$1"/>
>         <action application="set" data="proxy_media=false"/>
>         <action application="set" data="bypass_media=true"/>
>         </condition>
>     </extension>
>
> # sip_profiles/external.xml
>   <gateways>
>     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>   </gateways>
>   <settings>
>     <param name="disable-transcoding" value="false"/>
>     ...
>    </settings>
>
> # sip_profiless/external/example.xml
>         <gateway name="outbound1">
>                 <param name="realm" value="155.161.200.188:5070"/>
>                 <param name="register" value="false"/>
>                 <variables>
>                         <variable name="absolute_codec_string"
> value="PCMA" direction="outbound"/>
>                 </variables>
>                 <param name="reg-id" value="1"/>
>         </gateway>
>
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