Hello. I have a problem with recorded files (wav, mp3, gsm) there are three agents in callcenter which receives calls from same external provider. The problem is that recorded sound from one agent is good quality, while other two recorded sounds have dropped samples and its like the sound is at 2x speed. In vars.xml I have: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@32000h@20i,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> I am using mod_callcenter. There are 3 agents: user 892 at D-Link DPH-150S user 743 at D-Link DPH-150S user 897 at Linksys PAP2T There is incoming calls from same external number (multiple channels), the log is: ..... a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.658288 [DEBUG] sofia.c:6060 Remote SDP: a5079181-e072-4eee-9291-95a4ab40eabd v=0 a5079181-e072-4eee-9291-95a4ab40eabd o=- 1345205895 1345205895 IN IP4 192.168.0.6 a5079181-e072-4eee-9291-95a4ab40eabd s=- a5079181-e072-4eee-9291-95a4ab40eabd c=IN IP4 192.168.0.6 a5079181-e072-4eee-9291-95a4ab40eabd t=0 0 a5079181-e072-4eee-9291-95a4ab40eabd m=audio 13630 RTP/AVP 8 0 a5079181-e072-4eee-9291-95a4ab40eabd a=rtpmap:8 PCMA/8000 a5079181-e072-4eee-9291-95a4ab40eabd a=rtpmap:0 PCMU/8000 a5079181-e072-4eee-9291-95a4ab40eabd a=silenceSupp:off - - - - a5079181-e072-4eee-9291-95a4ab40eabd ..... a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[SPEEX:99:32000:20:44000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] sofia_glue.c:3027 Set Codec sofia/orange/extern_number@192.168.0.5:5061 PCMA/8000 20 ms 160 samples 64000 bits a5079181-e072-4eee-9291-95a4ab40eabd 2012-08-17 15:18:15.668301 [DEBUG] switch_core_codec.c:111 sofia/orange/extern_number@192.168.0.5:5061 Original read codec set to PCMA:8 After playing greeting, caller is routed to callcenter. When an agent answers the log has: For user 892: 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia.c:6060 Remote SDP: 55969678-ab9f-4007-8023-304f4b602c97 v=0 55969678-ab9f-4007-8023-304f4b602c97 o=710 38421252 24417309 IN IP4 132.101.208.92 55969678-ab9f-4007-8023-304f4b602c97 s=A conversation 55969678-ab9f-4007-8023-304f4b602c97 c=IN IP4 132.101.208.92 55969678-ab9f-4007-8023-304f4b602c97 t=0 0 55969678-ab9f-4007-8023-304f4b602c97 m=audio 10154 RTP/AVP 9 0 8 101 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:9 G722/16000 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:0 PCMU/8000 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:8 PCMA/8000 55969678-ab9f-4007-8023-304f4b602c97 a=rtpmap:101 telephone-event/8000 55969678-ab9f-4007-8023-304f4b602c97 a=fmtp:101 0-15 ...... 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[SPEEX:99:32000:20:44000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[G722:9:8000:20:64000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[PCMU:0:8000:20:64000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[PCMA:8:8000:20:64000] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [G722:9:16000:20:64000]/[GSM:3:8000:20:13200] 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:5097 Substituting codec G722@20i@8000h 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:892@132.101.208.92:5060 G722/8000 20 ms 160 samples 64000 bits 55969678-ab9f-4007-8023-304f4b602c97 2012-08-17 15:18:29.948337 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:892@132.101.208.92:5060 Original read codec set to G722:9 For user 743: acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia.c:6060 Remote SDP: acb5bf6a-4e79-46a4-a32b-420b5301d6ec v=0 acb5bf6a-4e79-46a4-a32b-420b5301d6ec o=1743 26961311 21406313 IN IP4 132.101.207.43 acb5bf6a-4e79-46a4-a32b-420b5301d6ec s=A conversation acb5bf6a-4e79-46a4-a32b-420b5301d6ec c=IN IP4 132.101.207.43 acb5bf6a-4e79-46a4-a32b-420b5301d6ec t=0 0 acb5bf6a-4e79-46a4-a32b-420b5301d6ec m=audio 10092 RTP/AVP 0 8 101 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:0 PCMU/8000 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:8 PCMA/8000 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=rtpmap:101 telephone-event/8000 acb5bf6a-4e79-46a4-a32b-420b5301d6ec a=fmtp:101 0-15 ...... acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[SPEEX:99:32000:20:44000] acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:743@132.101.207.43:5060 PCMU/8000 20 ms 160 samples 64000 bits acb5bf6a-4e79-46a4-a32b-420b5301d6ec 2012-08-17 15:20:29.918291 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:743@132.101.207.43:5060 Original read codec set to PCMU:0 For user 897: 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia.c:6060 Remote SDP: 35c3b541-c78e-4d4d-96fe-f15509dc7f91 v=0 35c3b541-c78e-4d4d-96fe-f15509dc7f91 o=- 1715257 1715257 IN IP4 132.101.202.2 35c3b541-c78e-4d4d-96fe-f15509dc7f91 s=- 35c3b541-c78e-4d4d-96fe-f15509dc7f91 c=IN IP4 132.101.202.2 35c3b541-c78e-4d4d-96fe-f15509dc7f91 t=0 0 35c3b541-c78e-4d4d-96fe-f15509dc7f91 m=audio 16444 RTP/AVP 0 100 101 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:0 PCMU/8000 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:100 NSE/8000 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=fmtp:100 192-193 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=rtpmap:101 telephone-event/8000 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=fmtp:101 0-15 35c3b541-c78e-4d4d-96fe-f15509dc7f91 a=ptime:30 ...... 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[SPEEX:99:32000:20:44000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5044 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:5097 Substituting codec PCMU@30i@8000h 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/sip:897@132.101.202.2:5060 PCMU/8000 30 ms 240 samples 64000 bits 35c3b541-c78e-4d4d-96fe-f15509dc7f91 2012-08-17 15:27:04.198310 [DEBUG] switch_core_codec.c:111 sofia/internal/sip:897@132.101.202.2:5060 Original read codec set to PCMU:0 See the difference for users 892 and 743. For user 892 a substitution for codecs is done, while for user 743 is not. Both users are registered from the same model of IP phone with same configs except sip configuration for login. What can be wrong? Thank you.
Forgot to mention. The user 743 has the good record audio file, the other two are bad recorded audio files.
So its always the same user who has good audio or does it matter who gets the call first? On Aug 17, 2012 12:46 PM, "Vbvbrj" <vbvbrj@gmail.com> wrote:
Forgot to mention. The user 743 has the good record audio file, the other two are bad recorded audio files.
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Sounds like network connection. Can you try switching stations between a bad phone and a good phone. Maybe your network has bad ports or cables somewhere. I know for all cisco voip they recommend cat 6 cables. On Aug 17, 2012 1:32 PM, "Vbvbrj" <vbvbrj@gmail.com> wrote:
On 17.08.2012 20:21, BookBag wrote:
So its always the same user who has good audio or does it matter who gets the call first?
The good record have the user 743 always, the other two always have bad sound record.
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I am very new to FreeSWITCH but I have been using it. Also I used Newfies and found the same problem. I just re sampled all my recordings to WAV in 8000Hz format and it fixed the problem. I do not know much about things but could this maybe be a codec problem in that your agents recordings that "work" are connecting to system with XXX codec yet your other agent that is "not working" are connecting with a different codec? Thanks, Obie Brown
On 17.08.2012 22:11, BookBag wrote:
Sounds like network connection. Can you try switching stations between a bad phone and a good phone. Maybe your network has bad ports or cables somewhere. I know for all cisco voip they recommend cat 6 cables.
It is definatelly not a network problem. The clients are connected to the same switch. I posted the log to see the difference. Observe "codec substitution" lines for bad clients. I don't know what this means to the system as why a substitution is done, but are not tried other codecs from client to compare.
In my DPH-150S for the problematic user in config at web management I switched "Signal Standard" from China to Russian, and this seems to solve the record problem. Why this is so influencing - I don't understand. How I will resolve the same issue for PAP2T?
participants (3)
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BookBag
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Obie Brown
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Vbvbrj