Trying to test mod_silk
I am trying to test mod_silk with FreeSWITCH rev. 17118. The problem appears to be that I haven't been able to get my FreeSWITCH to offer SILK as one of its codecs. mod_silk is loaded. I am using the PortAudio interface to make the call; the other end is running a recently built FreeSWITCH as well, thanks to Bill Acker for participating in the testing as always. I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with SILK@24000h as the first item in the comma-separated list. I have also tried it with SILK (no sampling rate specified). The SIP trace doesn't show that it is being offered. Bill has configured the other end in the same way. For example: a=rtpmap:10 L16/48000 a=rtpmap:114 CELT/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 is what appears as the SIP message from my local FreeSWITCH with SILK@24000h,CELT@48000h,G722,BV32,PCMU,PCMA,GSM in the codec preferences. (Actually, bv32 is out of date; I know the name changed and I should fix it - I haven't been playing with these settings for a while.) Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and IPv6 for making the connection. I know Brian and others have this working, so there's obviously something amiss here. As usual, there's no urgency - it's another attempt at a community contribution through testing.
Remove CELT. /b On Mar 26, 2010, at 10:56 PM, Jason White wrote:
I am trying to test mod_silk with FreeSWITCH rev. 17118.
The problem appears to be that I haven't been able to get my FreeSWITCH to offer SILK as one of its codecs.
mod_silk is loaded. I am using the PortAudio interface to make the call; the other end is running a recently built FreeSWITCH as well, thanks to Bill Acker for participating in the testing as always.
I have global_codec_prefs and outbound_codec_prefs in vars.xml configured with SILK@24000h as the first item in the comma-separated list. I have also tried it with SILK (no sampling rate specified).
The SIP trace doesn't show that it is being offered. Bill has configured the other end in the same way.
For example:
a=rtpmap:10 L16/48000 a=rtpmap:114 CELT/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000
is what appears as the SIP message from my local FreeSWITCH with SILK@24000h,CELT@48000h,G722,BV32,PCMU,PCMA,GSM in the codec preferences. (Actually, bv32 is out of date; I know the name changed and I should fix it - I haven't been playing with these settings for a while.)
Full SIP traces are available if desired. It's x86_64 Debian GNU/Linux and IPv6 for making the connection.
I know Brian and others have this working, so there's obviously something amiss here. As usual, there's no urgency - it's another attempt at a community contribution through testing.
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On 3/27/2010 7:22 PM, Jason White wrote:
Brian West<brian@freeswitch.org> wrote:
Remove CELT.
Unfortunately it didn't help.
a=rtpmap:10 L16/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
with the codes specified as SILK@24000h,G722,PCMU,PCMA
Try: silk@8000h@20i,G722@20i,PCMU@20i,PCMA@20i If it works, don't ask me why, just see: http://jira.freeswitch.org/browse/SFSIP-205 -- Andrew Thompson
Andrew Thompson <andrewkt@aktzero.com> wrote:
On 3/27/2010 7:22 PM, Jason White wrote:
Brian West<brian@freeswitch.org> wrote:
Remove CELT.
Unfortunately it didn't help.
a=rtpmap:10 L16/48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000
with the codes specified as SILK@24000h,G722,PCMU,PCMA
Try: silk@8000h@20i,G722@20i,PCMU@20i,PCMA@20i
Thank you for the suggestion; no luck. a=rtpmap:10 L16/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:10
On 3/27/2010 7:22 PM, Jason White wrote:
Brian West<brian@freeswitch.org> wrote:
Remove CELT.
Unfortunately it didn't help.
Please pastebin a full call attempt with console logging at 7. http://pastebin:freeswitch@pastebin.freeswitch.org/ -- Andrew Thompson
Andrew Thompson <andrewkt@aktzero.com> wrote:
Please pastebin a full call attempt with console logging at 7.
http://pastebin.freeswitch.org/12565 with SILK@24000h specified in my configuration as the only codec, and after having verified that mod_silk was loaded. All of the SILK codecs (8000, 12000, etc., including 24000) were logged as having been registered. I didn't notice anything interesting in the logs that I've posted other than the 48 khz codec being offered, presumably due to the fact that mod_portaudio was configured for 48 khz. Note that this always works perfectly well with a variety of codecs; FreeSWITCH resamples the input, as necessary, prior to encoding it.
Jason White <jason@jasonjgw.net> wrote:
http://pastebin.freeswitch.org/12565 with SILK@24000h specified in my configuration as the only codec, and after having verified that mod_silk was loaded. All of the SILK codecs (8000, 12000, etc., including 24000) were logged as having been registered.
I didn't notice anything interesting in the logs that I've posted other than the 48 khz codec being offered, presumably due to the fact that mod_portaudio was configured for 48 khz. Note that this always works perfectly well with a variety of codecs; FreeSWITCH resamples the input, as necessary, prior to encoding it.
Following a suggestion by Frank Carmickle, I confirmed that the profile status shows SILK as the first codec in both inbound and outbound lists. It still isn't showing up in the SIP request, though, so the problem remains exactly as described earlier in this thread.
participants (3)
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Andrew Thompson
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Brian West
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Jason White