A more complete debug-level log could be useful to get more context. Is box 2 running any other calls at the same time that might be using the license? Is box 2 doing anything else with the call? Anything like eavesdrop, recording, start_dtmf / start_dtmf_generate etc which uses the media will use a license, and I *think* that license only gets released until the end of the call. -Steve On 10 January 2013 07:09, Colin Mason <cmason@frontiernetworks.ca> wrote:
I have phone A connected to freeswitch box 1 and phone B connected to freeswitch box 2.
Phone A wants to dial phone B using G729. Codec is always G729.
The path RTP follows is:
Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B
(g729) (g729) (g729)
My question is, why is it that the freeswitch box receiving the call uses up an encoder/decoder when the codec is G729 along the path? If I reverse the call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license.
if I dial the PSTN to a carrier who supports G729, I don’t use up a license. Any thoughts? Maybe this is normal.
FreeSWITCH Box 1:
2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP [sofia/mpls/I888_1@172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 port 16466 codec: 18 ms: 20
2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 codec: 18 ms: 20
freeswitch@internal> g729_info
Permitted G729 channels: 40
Encoders in use: 0
Decoders in use: 0
FreeSWITCH Box 2:
2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/transport/3888@172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17 port 27276 codec: 18 ms: 20
2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP [sofia/mpls/sip:C996_1@10.253.200.10:5060] 172.16.16.16 port 18966 -> 10.253.200.10 port 16486 codec: 18 ms: 20
freeswitch@internal> g729_info
Permitted G729 channels: 40
Encoders in use: 1
Decoders in use: 1
Thanks in advance guys.
Colin
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