[Freeswitch-users] Inbound 1-way audio issue using GSM codec

[Anthony Minessale]

probably pstn side has acknowledged our gsm then sent ulaw anyway and we
think its gsm.
most likely there are multiple codecs in the accept packet from the gateway
and they expect us to figure out what codec to use based on the first packet
we get from them rather than just accepting one codec in the sdp like 90% of
devices so we have a proper chance to setup optimal packetization.  This is
one of those lame parts of the RFC that describe complete unscalable
stupidity that some stuff likes to tout for who knows why.

one thing you can try is to set the variable aboslute_codec_string in the
dial to force
only gsm to be advertised at all making it impossible for the remote end to
respond with multiple codecs.

<action application="bridge"
data="{absolute_codec_string=GSM}sofia/<profile>/<uri>"/>




On Fri, Nov 28, 2008 at 6:36 PM, Maxim Karp <mkarp at securesilence.com> wrote:

> Hello,
>
> I am using a GSM based endpoint connected to freeswitch that makes calls to
> the PSTN via a SIP gateway (SBC).  The SBC uses PCMU between itself and
> freeswitch.
>
> When I make an outgoing call from a GSM based device via freewsitch to the
> PSTN via the SBC, everything works fine and audio works in both directions
> for both end points.  I looked at the console logs and they do indicate
> that
> I am using GSM.
>
> Console output when I dial and before answer on the GSM device:
>
> v=0
> o=- 74 0 IN IP4 10.229.0.58
> s=session
> c=IN IP4 10.229.0.58
> b=CT:17
> t=0 0
> m=audio 59806 RTP/AVP 8 0 3 97 101
> k=base64:P6l1kBQy3canYTWZkxccjAVtTWO9g/N5L4gxLtX0UnM
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 RED/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=encryption:optional
>
> Console output once it rings and after I answer on the PSTN side:
>
> v=0
> o=FreeSWITCH 1227887572 1227887573 IN IP4 10.229.0.10
> s=FreeSWITCH
> c=IN IP4 10.229.0.10
> t=0 0
> a=sendrecv
> m=audio 30896 RTP/AVP 3 101 13
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> When I receive a call from the SIP gateway, the endpoint making the call
> (not on freeswitch) can't hear me speaking from the GSM device connected to
> freeswitch.  I can hear everything fine on the GSM device.
>
> Here is the console output for the call info coming in from the PSTN.
>
> v=0
> o=FreeSWITCH 1227902084 1227902085 IN IP4 38.113.164.132
> s=FreeSWITCH
> c=IN IP4 38.113.164.132
> t=0 0
> a=sendrecv
> m=audio 16724 RTP/AVP 0 101 13
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:13 CN/8000
> a=ptime:20
>
> Here is how I have vars.xml configured:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=GSM"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
>
> When I prioritize GSM on the outbound codec prefs I get static on the PSTN
> side.
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=GSM,PCMU,PCMA "/>
>
> Any ideas?
>
> Maxim.
>
>
> _______________________________________________
> 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
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081201/10ca2e22/attachment-0002.html