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IP Phones Bad Quality

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Bibbyboy

Bibbyboy

Programmer
Jan 26, 2009
70
US
I have a 406 Processor with 4.2.4 software on it, and I am using SIP trunking. The 5610sw ip phones that I have sound great ext to ext, but if you talk to somebody externally the person on the IP phone hears a terrible quality, however the person externally hears the person on the IP phone just fine. All the digital phones have great quality as well, to eliminate the IP network, I plugged the IP phone directly into the LAN port on the IP office and same result. We are using G7.11 ULAW 64k, and I went into Extensions and changed the IP phones to that instead of Automatic Selection as well as checked/unchecked Allow Direct Media Path and messed with some other check boxes. Any suggestions would be appreciated.
 
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It shouldnt make any difference as far as I know. They are both exactly the same in terms of hardware - the only difference is literally the number 1 or 2 lol
 
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I have a question concerning MattKnights comment about putting an IP Phone on the SIP Lan. When I do that the phone won't ever come up being that the TFTP server is on the voicemail machine which is a local address, how can I make this work?
 
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The phone will time out waiting for the TFTP server. after that it will register ok

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.
 
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MattKnight I have done that, but after it times out it gives me the classic "discover xxx.xxx.xxx.xxx" message x=voicemail IP
 
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172070mS CD: CALL: 277.1005.0 BState=Connected Cut=2 Music=0.0 Aend="SherryG(4123)" (0.0) Bend="Line 16" [Line 16] (0.0) CalledNum=2128574@74.223.147.141 () CallingNum=4123 (SherryG) Internal=0 Time=25682 AState=Idle
172071mS CD: CALL: 277.1005.0 Deleted
172071mS CMExtnEvt: SherryG: CALL LOST (CMCauseNormal)
172071mS CMExtnEvt: SherryG: Extn(4123) Calling Party Number(4123) Type(CMNTypeInternal)
172072mS CMExtnEvt: SherryG: CMExtnHandler::SetCurrent( id: 1005->0 )
172073mS CMCallEvt: 277.1005.0 -1 SherryG.-1: StateChange: END=X CMCSCompleted->CMCSDelete
172073mS CMExtnEvt: v=1 State, new=PortRecoverDelay old=CMESCompleted,0,0,SherryG
172075mS CMCallEvt: 16.1008.0 -1 SIPTrunk Endpoint: StateChange: END=X CMCSConnected->CMCSCompleted
172077mS SIP Tx: UDP 70.43.124.10:5060 -> 74.223.147.141:5060
BYE sip:2128574@74.223.147.141:5060 SIP/2.0
Via: SIP/2.0/UDP 70.43.124.10:5060;rport;branch=z9hG4bK9c9ebaf7aaa863267f11db880b48f1ad
From: "8034514123" <sip:8034514123@74.223.147.141>;tag=fa5a46fd2718d9a1
To: <sip:2128574@74.223.147.141>;tag=gK0cbd036b
Call-ID: 5ccd065b476cb6d9589250aaaddb5cb3@70.43.124.10
CSeq: 56639135 BYE
Contact: "8034514123" <sip:8034514123@70.43.124.10:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO
Content-Length: 0

172078mS CMCallEvt: END CALL:2 (fe9e9560)
172079mS PRN: Destroyed MH fe9e9ea8 parent unknown
172080mS CMCallEvt: 277.1005.0 -1 BaseEP: DELETE CMEndpoint fe9e9fb4 TOTAL NOW=1 CALL_LIST=0
172082mS H323Evt: SetRfc2833 (1): rx payload 101 tx payload 101
172082mS CMMap: PCG::UnmapBChan pcp[93]b0r1 cp_b 0 other_cp_b 0
172083mS CMMap: PCG::UnmapBChan pcp[91]b0r1 cp_b 0 other_cp_b 0
172083mS H323Evt: RTP(END): 192.168.1.240/49152 192.168.1.64/52274 CODEC 5 PKTSZ=160 RFC2833=off AGE=25673 SENT 698 RECV 542 RTdelay=128 jitter=2 loss=0 remotejitter=46 remoteloss=0
172092mS CD: CALLSYNC: cs02
172124mS SIP Rx: UDP 74.223.147.141:5060 -> 70.43.124.10:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 70.43.124.10:5060;branch=z9hG4bK9c9ebaf7aaa863267f11db880b48f1ad;rport=5060
From: "8034514123" <sip:8034514123@74.223.147.141>;tag=fa5a46fd2718d9a1
To: <sip:2128574@74.223.147.141>;tag=gK0cbd036b
Call-ID: 5ccd065b476cb6d9589250aaaddb5cb3@70.43.124.10
CSeq: 56639135 BYE
Content-Length: 0

172415mS RES: Thu 19/3/2009 13:31:03 FreeMem=43968300(1) CMMsg=5 (5) Buff=100 566 500 1108 5 Links=8047
172415mS RES2: RTEngine=0, CMRTEngine=0, Timer=58, Poll=0, Ready=0, CMReady=0, CMQueue=0, VPNNQueue=0
174073mS CMExtnEvt: SherryG: Recover Timer reason=CMTRWrapUp
174073mS CMExtnEvt: v=1 State, new=Idle old=PortRecoverDelay,0,0,SherryG
174074mS CMExtnTx: v=4123, p1=0
CMVoiceMailStatus
Line: type=IPLine 250 Call: lid=0 id=-1 in=0
Called[SherryG Msgs=1 Old=0 Sav=0] Type=Default (100) Reason=CMDRdirect Calling[00000001] Type=Default Plan=Default
Display [SherryG Msgs=1]
Timed: 19/03/09 13:31
177127mS CMCallEvt: 16.1008.0 -1 SIPTrunk Endpoint: StateChange: END=X CMCSCompleted->CMCSDelete
177128mS PRN: Destroyed MH fe9d9bc8 parent unknown
177129mS CMCallEvt: 16.1008.0 -1 BaseEP: DELETE CMEndpoint fe9da8f8 TOTAL NOW=0 CALL_LIST=0
177130mS H323Evt: SetRfc2833 (1): rx payload 101 tx payload 101
177131mS H323Evt: RTP(END): 70.43.124.10/49154 74.223.147.140/12652 CODEC 5 PKTSZ=160 RFC2833=on AGE=23197 SENT 515 RECV 653 RTdelay=0 jitter=0 loss=256 remotejitter=0 remoteloss=0
177414mS RES: Thu 19/3/2009 13:31:08 FreeMem=44004808(1) CMMsg=5 (5) Buff=100 568 500 1108 5 Links=8121
177415mS RES2: RTEngine=0, CMRTEngine=0, Timer=54, Poll=0, Ready=0, CMReady=0, CMQueue=0, VPNNQueue=0


I'm noticing the loss=256...does this help?
 
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>I have done that, but after it times out it gives me the classic "discover xxx.xxx.xxx.xxx" message x=voicemail IP


you need to give it an appropriate IP address, mask, Router and Gatekeeper address (Gatekeeper & Router) = IP address of LAN2 of IP406

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.
 
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Put the vm/tftp server ip address in the "Primary Trans. IP Adress" and enable NAT on LAN2. Then use the ip adress of LAN2 as the gatekeeper address for the IP Phones and use the ip address of the vm/tftp server as the fileserver ip address in the phones or move the vm server/tftp server to LAN2, or add a second NIC in the vm server and connect that to the LAN2, or add a router between LAN1 and LAN2 and use that as a gateway for the phones.
 
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I appreciate everyones help on this topic, as I am still having the issue. I have been on the phone with Avaya for the past couple days and they agree that I have everything right and that their is a codec mismatch. The carrier is telling me that they have their stuff set on G.711 ULAW, and I have my stuff set on that as well, if I tell them to set their stuff to G.729, and then I set the IP office to the same, (just for a mere test) is their any downsides? Is G.729 simply worse quality? What is the difference that I am looking at.
 
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Lower bit rate for quality 8kb of voice (total 35kb each way of traffic) where g711 uses 64kb for voice (same as isdn, total 95kb each way of traffic).

Most customers won't notice the difference.
 
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The fourth post in this tread mentioned a list of supported Avaya SIP trunk providers. Can anyone point me to where I can find this list?

Thanks.

V/
 
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Here you go vaniello

Greetzzz...Bas

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
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CODEC 5 PKTSZ=160 RFC2833=on AGE=27122 SENT 939 RECV 741 RTdelay=0 jitter=0 loss=256 remotejitter=0 remoteloss=0

what do these different things mean?
 
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loss=256, you have a local network problem.
So check the network/switches/ipoffice

Avaya_Red.gif

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________
 
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what about RTdelay? I have a lot of that on some of my other monitor files
 
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Just to let everyone know, I fixed the quality issue and it was indeed the carrier. Nuvox (our carrier) said that they talked with Advanced Services and they changed the packet service profile setting from "No Transcode" to "Nuvox Transcode", they said this was the first time they had done SIP Trunking with an Avaya phone system, and all their other customers have been set to "No Transcode", so if anyone ever uses Nuvox (at their own risk considering Avaya doesn't support it) make sure you tell them that. Thanx again everyone
 
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