Hi Peter, Thanks for your response. When I use PCMU two-way audio works fine. When I make outgoing calls from a Freeswitch extension (using GSM) and then out to a gateway using PCMU everything works fine. When I receive calls from the same gateway, the end point behind the gateway can't hear me. The GSM-PSMU (and viceversa) transcoding for outgoing from an endpoint associated with a Freeswitch extension to the external gateway is perfect but incoming there seems to be an issue. Maxim. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: December-01-08 3:39 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Inbound 1-way audio issue using GSM codec Hello Maxim, can you reach another internal device except the GSM one in order to see whether it's GSM codec specific? However I can see that you're using local IPs (10.x.x.x) so I expect that they are natted. This often causes one way audio when the external rtp-ip is not set. Please try to set a <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/> entry to internal.xml and external.xml in your SIP profiles and see if it works. Use stun at least for the internal profile (FQDN and external IP most probably will not work) Best regards Peter Maxim Karp schrieb:
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.
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