Can the GSM dongle do AMR? On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu < mishehu@freeswitch.org> wrote:
I think that AMR is one of the codecs that does not allow for transcoding, at least with the in-tree module. Might require something additional.
-- Yossi Neiman
On 04/07/2015 04:57 AM, Steven Ayre wrote:
That's not necessarily the case - GSM is not the only codec used by mobile networks, there's others such as AMR
On 4 April 2015 at 19:11, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Since one of the legs is GSM the best you can hope for is slightly worse than GSM because you are transcoding.
On Sat, Apr 4, 2015 at 2:01 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
for the freeswitch side <action application="set" data="absolute_codec_string=L16@16000h"/>
for the asterisk side you will have to figure that out yourself.
On Sat, Apr 4, 2015 at 12:40 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
thank you for reply it is freeswitch.conf <X-PRE-PROCESS cmd="set" data="sound_prefix=$${sounds_dir}/en/us/callie"/> <X-PRE-PROCESS cmd="set" data="default_country=US"/> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/>
<profile name="freeswitch-sip"> <gateways> <gateway name="asterisk-local"> <param name="proxy" value="127.0.0.1:5060"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> </gateway> </gateways>
<domains> <domain name="all" alias="true" parse="false"/> </domains>
<settings> <param name="debug" value="1"/> <param name="sip-trace" value="no"/> <param name="log-auth-failures" value="false"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="asterisk"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5050"/> <param name="dialplan" value="XML"/> <param name="dtmf-type" value="info"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="vad" value="none"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <param name="ext-rtp-ip" value="127.0.0.1"/> <param name="ext-sip-ip" value="127.0.0.1"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="false"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="auth-all-packets" value="false"/> <param name="challenge-realm" value="auto_from"/> </settings>
and here is asterisk sip.con disallow=all allow=ulaw allow=alaw
what i need to change ??
On Sat, Apr 4, 2015 at 9:13 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
probably the best you can do is limit the number of transcodings and/or resamplings
skype uses silk freeswitch core uses L16 sip session uses <your choice> asterisk core uses ? audio is presented to the dongle as ? cellular network uses gsm
if ? = L16 then make the sip session use L16 use the loopback interface so that MTU is not an issue
On Fri, Apr 3, 2015 at 2:50 PM, Shabbir abbasi < shabbirabbasi92@gmail.com> wrote:
for this setup skype --> freeswitch(mod_skypopen --> mod_sofia) ---> asterisk(chan_sip --> chan_dongle[Huawei E169] ) on same machine
suggest best codecs to use with freeswitch and asterisk for good voice quality ??
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