codec code used in SIP extension

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Is there anyway for me to find out what codec is used for my SIP extension as quality of voice among the extensions are very unsatisfactory (choppy, echo) even on the LAN segment which has only the 2 SIP extensions?

Thanks

Eric
 
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If u have problems inside the lan I think your problem will be in the server spec. It need higher processor.
If u have bandwidth problems I refer to ¨Elastix without teers¨ it explains how to activate g729 and g723.
 
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when you are in a call open a terminal enter the asterisk CLI and make a sip show channels , there you will be able to see the codecs your calls are using
 
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Noted from sip show channels, my sip channels are using alaw codec format. Is there anyway for me to change to a better codec for my sip channels?
 
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g711(ua) are the best "quality" codecs currently available in this distro for SIP, if you have choppy sound ext. to ext. on your LAN segment I don't believe that it is a codec problem. The question would be is the server robust enough and are you sure there is no problem with any shared interrupts?.

Is it a physical server running nothing but the basic services necessary for Elastix?
what does

cat /proc/interrupts

and

cat /proc/cpuinfo

show?
 
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As dicko saids , this may not be codec problem , because the problem is inside you LAN , not with remote peers , the problem is just between extensions? or you have the same problem with incoming calls?

Just to you to know i tell you how to change the codecs preferences, add your codec preferences in sip_general_custom.conf file , like this

videosupport=yes
disallow=all

allow=ulaw
allow=g729
allow=alaw
allow=h261
allow=h263
allow=h263p
language=en

dtmfmode=auto

The disallow option cancel the use of all codecs , then you put de allow codecs in the order you need them , in these example the asterisk will try to use ula fist and g729 next

Remember to config your extension to , after making a new extension enter and put there disallow all and in allow you can put the codecs in order like this allow=g729&ulaw , this is from the freepbx

Let me now your results
 
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The sip extension to sip extension voice quality was ok but I am having problems of choppy sound from incoming calls to the sip extensions. Here are the details that dicko needs to further diagnose my problems:

cat /proc/interrupts

CPU0 CPU1
0: 496495322 496493014 IO-APIC-edge timer
1: 1 2 IO-APIC-edge i8042
8: 2 1 IO-APIC-edge rtc
9: 0 0 IO-APIC-level acpi
12: 2 2 IO-APIC-edge i8042
50: 0 0 IO-APIC-level uhci_hcd:usb4
58: 0 0 IO-APIC-level uhci_hcd:usb6
66: 7552383 7551800 IO-APIC-level libata
74: 54269970 0 PCI-MSI eth0
169: 992948645 992951389 IO-APIC-level wcte12xp0, wcte12xp1
177: 0 0 IO-APIC-level uhci_hcd:usb3
225: 0 0 IO-APIC-level ehci_hcd:usb1
233: 99 93 IO-APIC-level ehci_hcd:usb2, uhci_hcd:usb5, uhci_hcd:usb7
NMI: 0 0
LOC: 994019317 994019306
ERR: 0
MIS: 0


and

cat /proc/cpuinfo

processor : 0
vendor_id : GenuineIntel
cpu family : 6
model : 23
model name : Intel(R) Xeon(R) CPU E3110 @ 3.00GHz
stepping : 10
cpu MHz : 2999.692
cache size : 6144 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 2
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 13
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni mon itor ds_cpl vmx smx est tm2 cx16 xtpr lahf_lm
bogomips : 6002.37

processor : 1
vendor_id : GenuineIntel
cpu family : 6
model : 23
model name : Intel(R) Xeon(R) CPU E3110 @ 3.00GHz
stepping : 10
cpu MHz : 2999.692
cache size : 6144 KB
physical id : 0
siblings : 2
core id : 1
cpu cores : 2
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 13
wp : yes
flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm constant_tsc pni mon itor ds_cpl vmx smx est tm2 cx16 xtpr lahf_lm
bogomips : 5999.28

BTW, asepulveda, I am a bit confused for your last paragraph:
"Remember to config your extension to , after making a new extension enter and put there disallow all and in allow you can put the codecs in order like this allow=g729&ulaw , this is from the freepbx".

Where can I input those parameters in freepbx as I did not see the relevant field there?


Regards,

Eric
 
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Well, it's not the interrupts or the hardware, I would suspect your WAN connection, is it big enough for your calls, are you are sharing it with any other amount of traffic, if so you will need a firewall that does QOS, have you found and used a bandwidth calculator, (there are couple here, just search for them.) if you appear restricted in any way then you are forced to start trading quality with choppiness as you go with a codec that does more and more compression (and if of course it is supported by your VSP.

I notice you have two digium 12x they seem to be sharing an interrupt and I seem to remember that there is "problem" with the pci-x versions, if so then a call to digium might be in order.

(found it : http://www.trixbox.org/forums/trixbox-f ... gium-cards)
 
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In the free pbx if you enter to an extension that you have configure you will see both fields , disallow and allow , here you must put disallow to all , and in allow the codecs in orden , separete the codecs with &

Your trunks are IP , Analogs or Digital (e1,t1)
 
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I am having a digital trunk, E1. As pointed out by dicko, I am having 2 Digium TE122B in a single box. My sip extension to sip extension was having a very good quality sound but choppy with incoming calls from the E1 trunk. BTW, I am running from a LAN, so basically I can rule out the WAN congestion.


Thanks


Eric
 
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Then the answer to your original post's question is without doubt "prefer" alaw, the E1 (european T1) will always use alaw. (g711a) so using anything else on your sip phones would be a waste of processing power and/or a resultant loss of quality.
Now I would go and investigate the apparent problems with 2 TE122's on the same machine as to interrupt/timing/stability. Can you test by removing one card, (you say you have one trunk but two E1 interfaces.)
 
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Hi dicko, I am having two E1 lines, each E1 card serving one E1 line. I cannot remove one of the card as they are using to serve live calls now. I just could not understand why 2 cards sharing the same interrupt will cause the sound degradation?

BTW, dicko, appreciate your assistance as you have been very helpful.

Eric
 
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Hi dicko,

Do you think memory of my server play a part in my choppy sound to the sip phone? The choppy sound problems for me is on and off (sometime got, sometime don't have). My server always recorded a 80% RAM utilisation (out of a 2GB RAM).


Eric
 
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top is a good tool, for showing what processes are using memory. Unless you are swapping a lot (also available in the top interface), it shouldn't be a problem. Yet something obviously is, are these digiums pci or pci-X?
 
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I believe in the link I sent you earlier there is a discussion of using 2x Digium pci-e cards where the end resolution was hardware replacement (by digium). I suggest you call the manufacturer to discuss your problem that might also be something to do with your other "s extension" problem if the D channel is also affected.
 
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Hi dicko, do you think that by having one PCI-e card and one using PCI card will solve the problem?


Thanks


Eric
 
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Personally, if In had a piece of hardware with any suspicion over it I would not use it, especially if I was handling 2 E1 worth of calls, again, I think that question is better asked of Digium,

Good luck.
 

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