[Freeswitch-users] How to configure g729 pass though on Freeswitch

David Ponzone david.ponzone at gmail.com
Mon Apr 5 03:02:27 PDT 2010


Frank,

again:

     
------------------------------------------------------------------------
recv 1101 bytes from udp/[192.168.4.154]:5060 at 20:41:04.437500:
     
------------------------------------------------------------------------
    INVITE sip:02074379497 at 192.168.4.156 SIP/2.0
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
    To: <sip:02074379497 at 192.168.4.156>
    Remote-Party-ID: Booth1 <sip: 
1001 at 192.168.4.156>;screen=yes;party=calling
    Call-ID: 68993a3f-e38f80af at 192.168.4.154
    CSeq: 102 INVITE
    Max-Forwards: 70
    Proxy-Authorization: Digest  
username="1001",realm="192.168.4.156",nonce="9567
e05b-d541-4a5e-9e47-2152eb90a199",uri="sip: 
02074379497 at 192.168.4.156",algorithm=
MD5 
,response 
="d0549f668825c7ce92e120071f1cb5ed",qop=auth,nc=00000001,cnonce="18d
a1954"
    Contact: Booth1 <sip:1001 at 192.168.4.154:5060>
    Expires: 240
    User-Agent: Linksys/SPA2102-3.2.8(d)
    Content-Length: 260
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
    Supported: x-sipura
    Content-Type: application/sdp

    v=0
    o=- 2612067 2612067 IN IP4 192.168.4.154
    s=-
    c=IN IP4 192.168.4.154
    t=0 0
    m=audio 16392 RTP/AVP 18 100 101
    a=rtpmap:18 G729a/8000
    a=rtpmap:100 NSE/8000
    a=fmtp:100 192-193
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:30
    a=sendrecv

     
------------------------------------------------------------------------
send 312 bytes to udp/[192.168.4.154]:5060 at 20:41:04.484375:
     
------------------------------------------------------------------------
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-36bfa4b3
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=3bb06cff17c3ecefo0
    To: <sip:02074379497 at 192.168.4.156>
    Call-ID: 68993a3f-e38f80af at 192.168.4.154
    CSeq: 102 INVITE
    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-17048M
    Content-Length: 0

     
------------------------------------------------------------------------
2010-04-02 21:41:04.593750 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1001 at 192.168.4.156 
  [cc4b83a6-3d16-4da3-bd6d-5148d0f983e8]
2010-04-02 19:06:27.500000 [NOTICE] sofia.c:4353 Hangup sofia/internal/1001 at 192.168.1.133 
  [CS_NEW] [INCOMPATIBLE_DESTINATION]

send 781 bytes to udp/[192.168.4.154]:5060 at 18:06:27.625000:
     
------------------------------------------------------------------------
    SIP/2.0 488 Not Acceptable Here
    Via: SIP/2.0/UDP 192.168.4.154:5060;branch=z9hG4bK-ec43882d
    From: Booth1 <sip:1001 at 192.168.4.156>;tag=bff390fd4255c0f9o0
    To: <sip:02074379497 at 192.168.4.156>;tag=m006c20Fg5Spa
    Call-ID: 1c27844d-e299e5e9 at 192.168.4.154

That is the answer from FS to the phone, just after receiving the  
INVITE that contains only G729 and NSE (??) in the SDP.

If you're sure you enabled G729 in vars.xml, you should check you  
restarted your SIP profile.
If you are not sure how to do it right, check my previous emails, or  
restart FS .

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pièces jointes sont confidentiels et établis  
à l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autorisée est interdite. Tout message électronique est  
susceptible d'altération. IPeva décline toute responsabilité au titre  
de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes  
pas destinataire de ce message, merci de le détruire immédiatement et  
d'avertir l'expéditeur.




Le 05/04/2010 à 11:49, Frank Church a écrit :

> Have you reviewed http://pastebin.freeswitch.org/12617 ?
> It has the G729 set in the codecs section.
> In this one it seems the call does not get to the external gateway.
> Freeswitch stops the call before calling the external gateway
>
>
> I have checked it again a few times using the bypass_media,
> proxy_media settings.
>
> And with those settings the call ends as soon as ringing starts or as
> sonn as the call is answered.
>
> I will do another one just to confirm
>
>
>
>
> On 4 April 2010 19:35, David Ponzone <david.ponzone at gmail.com> wrote:
>> No, FreeSWITCH does NOT need to be in bypass media or proxy media.
>> You just need the regular passthrough module: mod_g729 and to allow  
>> G729 as
>> inbound and outbound codecs in vars.xml.
>> To summarize:
>> -normal mode (proxy disabled, bypass disabled): FreeSWITCH is in  
>> the RTP
>> path, it relays the audio stream between endpoints, but can still  
>> detect
>> DTMFs
>> -proxy media enabled: FreeSWITCH relays the audio stream  
>> transparently, DTMF
>> detection is impossible. In this mode, FS is really a "dumb"  
>> transparent
>> RTP-forwarder (this is required to get T38 working between the 2  
>> endpoints)
>> -bypass media enabled: FreeSWITCH is not in the RTP path
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pièces jointes sont confidentiels et  
>> établis à
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autorisée est interdite. Tout message électronique est  
>> susceptible
>> d'altération. IPeva décline toute responsabilité au titre de ce  
>> message s'il
>> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire  
>> de ce
>> message, merci de le détruire immédiatement et d'avertir  
>> l'expéditeur.
>>
>>
>>
>> Le 04/04/2010 à 20:12, Max Clark a écrit :
>>
>> Clarification - for G729 does freeswitch need to be in "bypass media"
>> or "proxy media"? My understanding was that G729 would work with
>> "proxy media" enabled and without the new fangled module?
>>
>> -Max
>>
>> On Fri, Apr 2, 2010 at 2:23 AM, Jason White <jason at jasonjgw.net>  
>> wrote:
>>
>> Frank Church <vfclists at googlemail.com> wrote:
>>
>> I am just trialling Freeswitch with Linksys adapters, whose default
>>
>> codec I have set to G729 with 'Use Pref Codec Only:' set to no.
>>
>> When I change that setting to 'yes' the calls don't go through. I am
>>
>> using the latest Windows SVN.
>>
>> FreeSWITCH only supports G.729 in pass-through mode (i.e., with  
>> bypass media
>>
>> enabled). Apparently you can buy a G.729 licence for FreeSWITCH if  
>> you
>> really
>>
>> need it.
>>
>> My recommendation would be to use a codec other than G.729 unless  
>> you have a
>>
>> compelling reason, for example a carrier that only supports G.729.
>>
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>>
>> FreeSWITCH-users at lists.freeswitch.org
>>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>
>> http://www.freeswitch.org
>>
>>
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>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>>
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>>
>>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
> _______________________________________________
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