[Freeswitch-users] Speex transcoding question

[Bruce Hopkins]

OK - sorry to make such a meal of this.

I am still encountering a problem getting my configuration right, having
followed the advice I received previously.

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.

i.e. In vars.xml I have:

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=SPEEX at 16000h,SPEEX at 32000h
,SPEEX at 8000h,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>

 <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=SPEEX at 16000h
,SPEEX at 32000h,SPEEX at 8000h,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>


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.

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

Could anyone help with what I need to do to resolve my problem when
transcoding is required though please?

I am did "make current" last night so am using the latest snapshop.

many thanks again,
Bruce





On 16 March 2010 10:36, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:

> Please ignore this last email.  I think I have missed out something I hav
> already been told.  Sorry for the waste of bandwidth.
>
> regards
> Bruce
>
>
> On 16 March 2010 10:30, Bruce Hopkins <jbrucehopkins at gmail.com> wrote:
>
>> Hi,
>>
>> 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).
>>
>> If both phones use Speex wideband with no transcoding required by FS, the
>> call succeeds though.
>>
>> My codecs are listed in vars.xml as follows:
>>  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex at 32000
>> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex at 32000
>> @20i,speex at 16000h@20i,G722,G7221 at 32000h,G7221 at 16000h,PCMA,PCMU,GSM"/>
>>
>> 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) :
>>
>> Media Attribute (a): rtpmap:9 G722/8000
>> Media Attribute (a): rtpmap:99 SPEEX/8000
>> Media Attribute (a): rtpmap:115 G7221/32000
>> Media Attribute (a): fmtp:115 bitrate=48000
>> Media Attribute (a): rtpmap:107 G7221/16000
>> Media Attribute (a): fmtp:107 bitrate=32000
>> Media Attribute (a): rtpmap:8 PCMA/8000
>> Media Attribute (a): rtpmap:0 PCMU/8000
>> Media Attribute (a): rtpmap:3 GSM/8000
>> Media Attribute (a): rtpmap:101 telephone-event/8000
>> Media Attribute (a): fmtp:101 0-16
>> Media Attribute (a): rtpmap:13 CN/8000
>> Media Attribute (a): ptime:20
>>
>> Could anyone tell me what I am doing wrong please?
>>
>> Many thanks
>> Bruce
>>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100316/87695c47/attachment-0002.html