Hi Andreas,<div><br></div><div>Thank you very much. This is a good start for me. I also need to work with freeswitch. Not asterisk. And simple point to point sip is more than enough and rtp bridge would be fantastic.</div>
<div><br></div><div>Would it work with amr codec as well ? Or only gsm full rate/ h. Rate?<span></span></div><div><br></div><div>Im trying to find the lcr git location of jolly/new. Can you please give me the url. I cant find it on openbsc git.</div>
<div><br></div><div>Thanks again.</div><div><br></div><div>Nava.<br><br>On Tuesday, October 16, 2012, Andreas Eversberg  wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
hi kelum,<br>
<br>
in order to use sip, you need jolly/new branch of lcr and jolly/rtpmux of openbsc. the sip implementation of LCR is quite simple, so no authentication or oder features - just simple point-to-point SIP. if you run confiure of LCR, check if sip is enabled. in order to add a SIP interface, do the following at interface.conf:<br>

<br>
[sip]<br>
sip &lt;local ip&gt;[:&lt;local port&gt;] &lt;remote ip&gt;[:&lt;remote port&gt;]<br>
earlyb yes<br>
tones no<br>
<br>
you need to define local IP that will be used to connect to remote SIP endpoint. don&#39;t use localhost, if the endpoint is on a different machine, because this IP is also used for RTP. if you use same machine, you need to have different ports. you may change local port, by adding a local port or you may change port of SIP endpoint and then add remote port.<br>

<br>
i have tested it with freeswitch, but asterisk should work also.<br>
<br>
you may then also try at interface.conf below &quot;[sip]&quot; definition:<br>
<br>
rtp-brige<br>
<br>
then the codec (full rate or enhanced full rate) is negotiated between mobile and the remote SIP endpoint. the SIP endpoint must support at least one of it.<br>
<br>
regards,<br>
<br>
andreas<br>
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</blockquote></div>