Hello Daniel ,<br><br>I'm Also Having the Same doubt on g729 Codec,<br><br>I'm using RTPproxy with Nathelper , <br>OpenSER and RTP proxy does media signaling when the Call is Established, <br><br>My main question is
<br>Is RTP proxy support the G729, with OpenSER, <br><br>With out using the Transcoder ( Asterisk ) How can OpenSER signals the G729 Codec.<br><br><div><span class="gmail_quote">On 2/15/07, <b class="gmail_sendername">Daniel-Constantin Mierla
</b> <<a href="mailto:daniel@voice-system.ro">daniel@voice-system.ro</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br><br>you need a transcoder in the middle. OpenSER does only signaling, so it<br>is not able to transcode. Asterisk, for example, does.<br><br>Cheers,<br>Daniel<br><br>On 02/15/07 10:57, tusker keg wrote:<br>> Howdy
<br>><br>> I have a situation I hope you guys will help me out with<br>><br>> I am receiving call from a VOIP peer (SIP Call) and the peer can<br>> only send them as G711.<br>> I need to redirect to call to another voip peer over the wan and due
<br>> to bandwidth considerations I need to translate the codec to g729.<br>><br>> Any ideas on how to do this,<br>><br>> Sample config file will be help<br>><br>> Regards<br>><br>> ./Tusker<br>
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