vicidial.org

Hi Vicidial users,

While this CLI notice is flooding, agents have problem with conversations (not clear, metalic voice):

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Nov  6 14:38:48 NOTICE[7376]: rtp.c:587 ast_rtp_read: Unknown RTP codec 90 received
Nov  6 14:38:48 NOTICE[7376]: rtp.c:587 ast_rtp_read: Unknown RTP codec 90 received
Nov  6 14:38:48 NOTICE[7376]: rtp.c:587 ast_rtp_read: Unknown RTP codec 90 received

This notice is flood ~once/hour.

I got also:

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Nov  6 14:43:10 NOTICE[12007]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.11.128.50

This information appears several times a minute.

load average: 0.25, 0.24, 0.19

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VICIDIAL: Server Performance                             2008-11-06 14:47:11
Time range: 2008-11-06 09:00:00 to 2008-11-06 15:30:00

---------- TOTALS, PEAKS and AVERAGES
Total Calls in/out on this server:              3426
Total Off-Hook time on this server (min):    1441.65
Average/Peak channels in use for server:     32.5661 / 70
Average/Peak load for server:                22.4300 / 120
Average USER process cpu percentage:          3.2103 %
Average SYSTEM process cpu percentage:        1.0249 %
Average IDLE process cpu percentage:         95.8617 %


Total working agents: 6-8

All conversations are recording. Agents uses Zoiper (IAX)

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CLI> iax2 show channels
vici*CLI> iax2 show channels
Channel               Peer             Username    ID (Lo/Rem)  Seq (Tx/Rx)  Lag      Jitter  JitBuf  Format
IAX2/cc318-1          192.168.17.48    (None)      00001/00797  00020/00021  00000ms  -0001ms  0000ms  ulaw
IAX2/cc315-9          192.168.17.45    (None)      00009/00905  00027/00028  00000ms  -0001ms  0000ms  ulaw
IAX2/cc317-13         192.168.17.47    (None)      00013/00229  00006/00007  00000ms  -0001ms  0000ms  ulaw
IAX2/cc311-17         192.168.17.41    (None)      00017/00799  00016/00017  00000ms  -0001ms  0000ms  ulaw
IAX2/cc320-21         192.168.17.49    (None)      00021/00922  00022/00023  00000ms  -0001ms  0000ms  ulaw
IAX2/cc310-23         192.168.17.40    (None)      00023/00936  00014/00015  00000ms  -0001ms  0000ms  ulaw
6 active IAX channels


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 sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
194.9.25.21      0048862726  05ec3d8a643  00103/00000  ulaw  No       Tx: INVITE
194.9.25.21      0048655299  5823a919147  00103/00000  ulaw  No       Tx: INVITE
194.9.25.21      0048598231  53174988643  00103/00000  ulaw  No       Tx: ACK
194.9.25.21      gsgcall3    0dd5d98f0a5  00145/00000  unkn  No
194.9.25.21      gsgcall8    018cb3ac7c6  00145/00000  unkn  No
194.9.25.21      gsgcall6    575452da22d  00145/00000  unkn  No
194.9.25.21      gsgcall5    14920e6c2c7  00145/00000  unkn  No
194.9.25.21      gsgcall9    6a7f3a6c040  00145/00000  unkn  No
194.9.25.21      gsgcall1    000ec20d650  00145/00000  unkn  No
194.9.25.21      gsgcall     042889f13cf  00145/00000  unkn  No
194.9.25.21      gsgcall2    7fc39b34033  00145/00000  unkn  No
194.9.25.21      gsgcall4    57973a98693  00145/00000  unkn  No
194.9.25.21      gsgcall7    75488490543  00145/00000  unkn  No
194.9.25.21      0048768620  33b27ead562  00103/00000  ulaw  No       Tx: ACK
194.9.25.21      0048857404  13c6c3d111e  00103/00000  ulaw  No       Tx: ACK
194.9.25.21      0048768466  7539263c310  00103/00000  ulaw  No       Tx: ACK
16 active SIP channels


Any sugestion?

hi we have the same problem....i also had that voice are not clear when the rtp codec error appears on CLI mode in asterisk....hope anyone can explain & answer this...

Dear Matt,

Can You take a look for this?

RTP debugging is not my area of expertise. Using most mainstream VOIP providers on a network with a good firewall I never run into these kinds of issues.

This is caused by your provider requesting an audio codec that your asterisk installation is unable to support. This is either caused by lack of licenses for the codec or because Asterisk just does not have it implemented. Contact your provider and find out which audio codec they support and make sure you have only those codecs allowed in your sip.conf file.

As for the comfort noise issue check if your provider can disable this on your account. If not it should not cause an issue, but it is better to have it off if possible.