OK - sorry to make such a meal of this. <br><br>I am still encountering a problem getting my configuration right, having followed the advice I received previously.<br><br>What I am observing is that if more than one flavour of SPEEX is enabled in vars.xml, then only the first one in the list is offered to phone B in the SDP section of the INVITE if transcoding is required by Freeswitch.<br>
<br>i.e. In vars.xml I have:<br><br><X-PRE-PROCESS cmd="set" data="global_codec_prefs=SPEEX@16000h,SPEEX@32000h,SPEEX@8000h,G722,G7221@32000h,G7221@16000h,PCMA,PCMU,GSM"/><br><br> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=SPEEX@16000h,SPEEX@32000h,SPEEX@8000h,G722,G7221@32000h,G7221@16000h,PCMA,PCMU,GSM"/><br>
<br><br>If I then try to make a call from Phone A (using PCMA) to Phone B (using SPEEX/8000), the call fails and the log shows an incompatible destination. The SDP of the INVITE from Freeswitch to Phone B offers SPEEX/16000 but not SPEEX/8000 or SPEEX/32000, so there is no compatible codec offered to Phone B.<br>
<br>If however both Phone A and Phone B both use SPEEX/8000, the call works fine - as my default configuration is that the codec preferred by Phone A goes to the top of the list of codecs offered to Phone B<br><br>Could anyone help with what I need to do to resolve my problem when transcoding is required though please?<br>
<br>I am did "make current" last night so am using the latest snapshop.<br><br>many thanks again,<br>Bruce<br><br><br><br><br><br><div class="gmail_quote">On 16 March 2010 10:36, Bruce Hopkins <span dir="ltr"><<a href="mailto:jbrucehopkins@gmail.com">jbrucehopkins@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Please ignore this last email. I think I have missed out something I hav already been told. Sorry for the waste of bandwidth.<br>
<br>regards<br><font color="#888888">Bruce</font><div><div></div><div class="h5"><br><br><div class="gmail_quote">On 16 March 2010 10:30, Bruce Hopkins <span dir="ltr"><<a href="mailto:jbrucehopkins@gmail.com" target="_blank">jbrucehopkins@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi,<br><br>I am having trouble getting my configuration right so that I can have a call transcoded to Speex wideband from another codec (alaw or g.722).<br>
<br>If both phones use Speex wideband with no transcoding required by FS, the call succeeds though.<br>
<br>My codecs are listed in vars.xml as follows:<br> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@32000@20i,speex@16000h@20i,G722,G7221@32000h,G7221@16000h,PCMA,PCMU,GSM"/><br><br><X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@32000@20i,speex@16000h@20i,G722,G7221@32000h,G7221@16000h,PCMA,PCMU,GSM"/><br>
<br>However if I make a call from a phone A using, say, g.722 to a phone using Speex wideband, the SDP in the invite from phone A to phone B does not include Speex wideband. In fact the SDP includes speex/8000 even though speex/8000 is neither enabled in vars.xml, not in either of the phones. Here is the codec listing in the SDP of the INVITE from Freeswitch to phone B (from Wireshark) :<br>
<br>Media Attribute (a): rtpmap:9 G722/8000<br>Media Attribute (a): rtpmap:99 SPEEX/8000<br>Media Attribute (a): rtpmap:115 G7221/32000<br>Media Attribute (a): fmtp:115 bitrate=48000<br>Media Attribute (a): rtpmap:107 G7221/16000<br>
Media Attribute (a): fmtp:107 bitrate=32000<br>Media Attribute (a): rtpmap:8 PCMA/8000<br>Media Attribute (a): rtpmap:0 PCMU/8000<br>Media Attribute (a): rtpmap:3 GSM/8000<br>Media Attribute (a): rtpmap:101 telephone-event/8000<br>
Media Attribute (a): fmtp:101 0-16<br>Media Attribute (a): rtpmap:13 CN/8000<br>Media Attribute (a): ptime:20<br><br>Could anyone tell me what I am doing wrong please?<br><br>Many thanks<br><font color="#888888">Bruce<br>
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