Thank you very andreas, for the help and nice integration work.<div><br></div><div>Im going to try this branch now.</div><div><br></div><div>Is it only this patch to be applied to the rtp_proxy.c for the git clone i have from openbsc ?</div>
<div><br></div><div><a href="http://cgit.osmocom.org/cgit/openbsc/commit/?h=jolly/rtpmux">http://cgit.osmocom.org/cgit/openbsc/commit/?h=jolly/rtpmux</a> </div><div><br><br><div class="gmail_quote">On Thu, Mar 8, 2012 at 8:09 AM, jolly <span dir="ltr">&lt;<a href="mailto:andreas@eversberg.eu">andreas@eversberg.eu</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">hi nik,<br>
<br>
since lcr uses alaw (optionally ulaw) internally, it transcodes the<br>
audio between alaw and gsm full rate.<br>
<br>
alternatively you can bridge rtp directly between freeswitch and<br>
openbsc, if you use the &quot;bridge &lt;interface&gt;&quot; feature at interface.conf.<br>
additionally you need to add a line &quot;rtp-bridge&quot; to your interface<br>
definition. this rtp-bridge feature only works with the bridge feature,<br>
since lcr will not be able to handle audio streams then. since it is<br>
still developed, you need to use the &quot;jolly/rtpmux&quot; branch of openbsc.<br>
then it will only offer the codec to freeswitch which the phone<br>
supports. on the other direction, lcr selects the commonly supported<br>
codec. if more codecs are supported, the upper most (prefered) is used.<br>
<br>
anyway it should work in both cases. (with and  without rtp-bridge feature)<br>
<br>
regards,<br>
<br>
andreas<br>
<br>
</blockquote></div><br></div>