<div dir="ltr">Hello, Shlomi<div><br><div><div>RTP voice through WiFi is not so good, but it depends on settings and currently available bandwith of WiFi routers, your and neighbors.</div><div>I have ARM device with Freeswitch and mod_gsmopen and should said without WiFi in the voice path it work much better.</div>
<div>I have checked a lot of SIP-UA and codecs. Currently I use g.729a, as it work slightly better.</div><div>This problem can be decreased by increasing jitter buffer size, but in this case delay in voice can be too large. (GSM part already have noticeable latency)</div>
</div></div><div><br></div><div>Best Wishes,</div><div> Dmitry.</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-06-18 18:11 GMT+03:00 Shlomi Agiv <span dir="ltr"><<a href="mailto:shlomi.agiv@cell-buddy.com" target="_blank">shlomi.agiv@cell-buddy.com</a>></span>:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi <span style="font-family:arial,sans-serif;font-size:12.727272033691406px">Giovanni,</span><div><span style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br>
</span></div><div><span style="font-family:arial,sans-serif;font-size:12.727272033691406px">First let me say it's an honor for me to hear from you.</span></div>
<div><span style="font-family:arial,sans-serif;font-size:12.727272033691406px">As someone who has done a lot of coding using(and also modifying a bit) the gsmopen, talking to the author feel to me like talking to a celebrity.</span></div>
<div><font face="arial, sans-serif"><br></font></div><div><font face="arial, sans-serif">So far, we've connected the gsmopen and managed to make and receive calls, send and receive sms, which work well.</font></div><div>
<font face="arial, sans-serif">We haven't tested on wifi yet, should be done next.</font></div><div><font face="arial, sans-serif"><br></font></div><div><font face="arial, sans-serif">Two issues we haven't gotten around to debug, maybe you can help with:</font></div>
<div><font face="arial, sans-serif"><b>ussd:</b></font></div><div><font face="arial, sans-serif">sending ussd always returns the text "just a ussd message" whatever i send</font></div><div><span style="font-family:arial,sans-serif"><br>
</span></div><div><span style="font-family:arial,sans-serif"><b>NORMAL_CIRCUIT_CONGESTION:</b></span><font face="arial, sans-serif"><br></font></div><div><font face="arial, sans-serif">on occasion, when trying to make a call, we're getting 'NORMAL_CIRCUIT_CONGESTION' and we're not getting any incoming calls.</font></div>
<div><font face="arial, sans-serif">Currently what we do is unload mod_gsmopen, remove and reinsert the usb stick, then load mod_gsmopen. </font></div><div><span style="font-family:arial,sans-serif">This also doesn't always help since sometimes we can't unload the gsmopen module (it says 'model in use'). in that case we restart freeswitch and reinsert the usb stick.</span></div>
<div><font face="arial, sans-serif">This might not be caused by the gsmopen, since we're controlling the messages coming from the sim, and we might be doing something wrong.</font></div><div><br></div><div><font face="arial, sans-serif">Thanks,</font></div>
<div><font face="arial, sans-serif">Shlomi Agiv</font></div><div><span style="font-family:arial,sans-serif;font-size:12.727272033691406px"><br></span></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra">
<br><br><div class="gmail_quote">On Wed, Jun 18, 2014 at 4:29 PM, Giovanni Maruzzelli <span dir="ltr"><<a href="mailto:gmaruzz@gmail.com" target="_blank">gmaruzz@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><div>gsm voice quality will be good<br><br></div>Codec choice is only relevant for the SIP leg, so use your normal choice in term of bandwidth/cpu/ptime.<br>
<br></div>Gsmopen does not use codec, it runs internally in raw signed linear format.<br>
<br></div>Let us (or me personally) know how the project goes.<br><br></div>-giovanni<br><br><br></div><div class="gmail_extra"><br clear="all"><div>Sincerely,<br><br>Giovanni Maruzzelli<br>Cell : <a href="tel:%2B39-347-2665618" value="+393472665618" target="_blank">+39-347-2665618</a><br>
</div>
<br><br><div class="gmail_quote"><div><div>On Wed, Jun 18, 2014 at 3:12 PM, Shlomi Agiv <span dir="ltr"><<a href="mailto:shlomi.agiv@cell-buddy.com" target="_blank">shlomi.agiv@cell-buddy.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>We intend to put freeswitch on an arm powered portable device, which will connect using a usb gsm stick and will serve as a wifi/gsm router<br></div><div><br></div><div>so, a call will connect two endpoints - one to the sip client through wifi, the other through mod_gsmopen to the gsm.</div>
<div><br></div><div>How will the voice quality be? which voice codec is most recommended? </div><div><br></div><div><br></div></div>
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