<div dir="ltr">Can the GSM dongle do AMR?</div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 7, 2015 at 12:42 PM, I put the Who? in Mishehu <span dir="ltr"><<a href="mailto:mishehu@freeswitch.org" target="_blank">mishehu@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<div>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.<span class="HOEnZb"><font color="#888888"><br>
<br>
<pre cols="72">--
Yossi Neiman
On 04/07/2015 04:57 AM, Steven Ayre wrote:
</pre>
</font></span></div><div><div class="h5">
<blockquote type="cite">
<div dir="ltr">That's not necessarily the case - GSM is not the
only codec used by mobile networks, there's others such as <span style="color:rgb(37,37,37);font-family:sans-serif;font-size:14px;line-height:22.3999996185303px">AMR</span></div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On 4 April 2015 at 19:11, Russell
Treleaven <span dir="ltr"><<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Since one of the legs is GSM the best you can
hope for is slightly worse than GSM because you are
transcoding.
<div><br>
</div>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Sat, Apr 4, 2015 at 2:01
PM, Russell Treleaven <span dir="ltr"><<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">for the freeswitch side
<div><action application="set"
data="absolute_codec_string=L16@16000h"/></div>
<div><br>
</div>
<div>for the asterisk side</div>
<div>you will have to figure that out yourself.</div>
<div><br>
<div><br>
</div>
</div>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Sat, Apr 4, 2015
at 12:40 PM, Shabbir abbasi <span dir="ltr"><<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>
<div>
<div>thank you for reply<br>
</div>
it is freeswitch.conf<br>
<X-PRE-PROCESS cmd="set"
data="sound_prefix=$${sounds_dir}/en/us/callie"/><br>
<X-PRE-PROCESS cmd="set"
data="default_country=US"/><br>
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br>
<br>
<profile
name="freeswitch-sip"><br>
<gateways><br>
<gateway
name="asterisk-local"><br>
<param
name="proxy" value="<a href="http://127.0.0.1:5060" target="_blank">127.0.0.1:5060</a>"/><br>
<param
name="retry-seconds"
value="30"/><br>
<param
name="caller-id-in-from"
value="true"/><br>
</gateway><br>
</gateways><br>
<br>
<domains><br>
<domain name="all"
alias="true" parse="false"/> <br>
</domains><br>
<br>
<settings><br>
<param name="debug"
value="1"/><br>
<param
name="sip-trace" value="no"/><br>
<param
name="log-auth-failures"
value="false"/><br>
<param
name="forward-unsolicited-mwi-notify"
value="false"/><br>
<param
name="context"
value="asterisk"/><br>
<param
name="rfc2833-pt" value="101"/><br>
<param
name="sip-port" value="5050"/><br>
<param
name="dialplan" value="XML"/><br>
<param
name="dtmf-type" value="info"/><br>
<param
name="inbound-codec-prefs"
value="$${global_codec_prefs}"/><br>
<param
name="outbound-codec-prefs"
value="$${global_codec_prefs}"/><br>
<param
name="use-rtp-timer"
value="true"/><br>
<param
name="rtp-timer-name"
value="soft"/><br>
<param
name="rtp-timeout-sec"
value="300"/><br>
<param
name="rtp-hold-timeout-sec"
value="1800"/><br>
<param name="vad"
value="none"/> <br>
<param
name="rtp-ip"
value="127.0.0.1"/><br>
<param
name="sip-ip"
value="127.0.0.1"/><br>
<param
name="ext-rtp-ip"
value="127.0.0.1"/><br>
<param
name="ext-sip-ip"
value="127.0.0.1"/><br>
<param
name="inbound-codec-negotiation"
value="generous"/><br>
<param name="tls"
value="false"/><br>
<param
name="nonce-ttl" value="60"/><br>
<param
name="auth-calls"
value="false"/><br>
<param
name="auth-all-packets"
value="false"/><br>
<param
name="challenge-realm"
value="auto_from"/><br>
</settings><br>
<br>
<br>
</div>
and here is asterisk sip.con<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
<br>
</div>
what i need to change ??<br>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Sat,
Apr 4, 2015 at 9:13 PM, Russell
Treleaven <span dir="ltr"><<a href="mailto:rtreleaven@bunnykick.ca" target="_blank">rtreleaven@bunnykick.ca</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>probably the best you
can do is limit the number
of transcodings and/or
resamplings</div>
<div><br>
</div>
<div>
<div>skype uses silk</div>
<div><span style="white-space:pre-wrap">
</span>freeswitch core
uses L16</div>
<div><span style="white-space:pre-wrap">
</span>sip session uses
<your choice></div>
<div><span style="white-space:pre-wrap">
</span>asterisk core
uses ?</div>
<div><span style="white-space:pre-wrap">
</span>audio is
presented to the dongle
as ?</div>
<div><span style="white-space:pre-wrap">
</span>cellular network
uses gsm</div>
<div><br>
</div>
<div>if ? = L16 then make
the sip session use L16</div>
<div>use the loopback
interface so that MTU is
not an issue</div>
</div>
<div><br>
</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">
<div>
<div>On Fri, Apr 3, 2015
at 2:50 PM, Shabbir
abbasi <span dir="ltr"><<a href="mailto:shabbirabbasi92@gmail.com" target="_blank">shabbirabbasi92@gmail.com</a>></span>
wrote:<br>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div>
<div dir="ltr">
<div>for this
setup<br>
</div>
skype -->
freeswitch(mod_skypopen
--> mod_sofia)
--->
asterisk(chan_sip
-->
chan_dongle[Huawei
E169] ) on
same machine <br>
<br>
suggest best
codecs to use with
freeswitch and
asterisk for good
voice quality ??<br>
</div>
<br>
</div>
</div>
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