No Voice is streaming from the VOIP telephone to outside.
Hi,
I have a Cisco UC520, the SIP trunk is configured and it is up, the incoming calls are transfered to the approperiate extension and users from internal can make a call to outside. when the users call to internal extensions they can talk nice. but when they make a call to external or somebody from outside make a call to the internal the voice from users are not streamed to the other party.
I check the UC access-list and RTP ports are open (UDP from 16384 to 32767) but still can not heared from out side.
totally the call is initiated but the voice just come into the users and when the users speak on the phone the other party can not hear anything.
for more information, the UC device is NAT device for users and it gvies the IP to the clients and phones. it is behind a ADSL router and the internal ip of the ADLS is the default gateway of the UC. so the users will come though the UC and then go to the modem and then to the internet.
any idea about the voice problem please.
I have a Cisco UC520, the SIP trunk is configured and it is up, the incoming calls are transfered to the approperiate extension and users from internal can make a call to outside. when the users call to internal extensions they can talk nice. but when they make a call to external or somebody from outside make a call to the internal the voice from users are not streamed to the other party.
I check the UC access-list and RTP ports are open (UDP from 16384 to 32767) but still can not heared from out side.
totally the call is initiated but the voice just come into the users and when the users speak on the phone the other party can not hear anything.
for more information, the UC device is NAT device for users and it gvies the IP to the clients and phones. it is behind a ADSL router and the internal ip of the ADLS is the default gateway of the UC. so the users will come though the UC and then go to the modem and then to the internet.
any idea about the voice problem please.
IanTh
have you discussed this with cisco are you running the latest version? What protocol are you using ?
ASKER
the UC is using the Cisco IOS 8:6 and sip trunk is configured for sip and g711ulaw and secondary option of g729.
because all the port from inside is open I am wondering why the rtp is not going.
because all the port from inside is open I am wondering why the rtp is not going.
yes thats why I said have you discussed it with cisco I cannot see why its not working and it could be a known issue
ASKER
I have debug the UC when I am calling and I found the following error code:
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 488
I check the negociated codec and it was g729r8 in the log, so I added the g729r8 codec to the voice class codec 1. but still I have a problem. now even the call is not coming to the system. it seems that what I have changed to troubleshoot the voice streaming caused a bigger problem. now I am able just to call to our side and the voice is not stream to outside.
Disconnect Cause (CC) : 65
Disconnect Cause (SIP) : 488
I check the negociated codec and it was g729r8 in the log, so I added the g729r8 codec to the voice class codec 1. but still I have a problem. now even the call is not coming to the system. it seems that what I have changed to troubleshoot the voice streaming caused a bigger problem. now I am able just to call to our side and the voice is not stream to outside.
sip 488
488 Not Acceptable Media
This response indicates an error in handling the request at this time.
The SIP gateway generates this response if the media negotiation fails.
cc 65
Media Negotiation Failure
Typical scenarios include:
•No codec match occurred.
•H.323 or H.245 problem leading to failure in media negotiation.
65
CC_CAUSE_BEARER_CAPABILITY
Indicates that the equipment sending this cause does not support the bearer capability requested.
488 Not Acceptable Media
This response indicates an error in handling the request at this time.
The SIP gateway generates this response if the media negotiation fails.
cc 65
Media Negotiation Failure
Typical scenarios include:
•No codec match occurred.
•H.323 or H.245 problem leading to failure in media negotiation.
65
CC_CAUSE_BEARER_CAPABILITY
Indicates that the equipment sending this cause does not support the bearer capability requested.
ASKER
IanTh thanks for your reply, you are right, I changed the codec of dial-peer to g711ulaw and now I am getting the ring and now error is generated.
I am waiting for tomorrow morning to see if the voice is going out of the company. I will inform you tomorrow.
I am waiting for tomorrow morning to see if the voice is going out of the company. I will inform you tomorrow.
ASKER
unfortunately the voice is not streamed to the other party on telephone talk. Any idea please
My firts thought would be "NAT breaks the communication."
What IP phone models are you using? and can you please attach the output from a debug ccsip messages from during a test call and add the calling and called numbers, and a description of the IPs involved?
THanks,
What IP phone models are you using? and can you please attach the output from a debug ccsip messages from during a test call and add the calling and called numbers, and a description of the IPs involved?
THanks,
ASKER
I make a call to number 0290788356, you can see the log at attached. (i enabled the debug ccsip calls), for more info, 172.16.1.10 is the IP address of UC which is going to 172.16.1.1 (IP address of ADSL modem/Default gateway). log.txt
then I also enabled the ccsip messages and the file message-log.txt contains the complete log.
in both logs, I called and somebody from company pickedup the phone and I couldn't hear his/her voice.
IP phones are Cisco 7940, there is no problem with IP phones while they are able to call to internal extensions and the problem is just when the call is from/to outside.
thanks
Log.txt
message-log.txt
then I also enabled the ccsip messages and the file message-log.txt contains the complete log.
in both logs, I called and somebody from company pickedup the phone and I couldn't hear his/her voice.
IP phones are Cisco 7940, there is no problem with IP phones while they are able to call to internal extensions and the problem is just when the call is from/to outside.
thanks
Log.txt
message-log.txt
ASKER CERTIFIED SOLUTION
membership
This solution is only available to members.
To access this solution, you must be a member of Experts Exchange.
Ok, great to hear that. I am wondering how your ADSL router or even your provider handles SIP, reason being, the UC520 tells the other end to send the audio to an internal IP:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.161.164.69:5060;branch
From: "Anonymous"<sip:anonymous@
To: "Patrick Hall"<sip:0290788356@voice
Date: Tue, 10 Jul 2012 23:48:57 GMT
Call-ID: SDtktfe01-602dd73951c091f2
CSeq: 967170713 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:0290788356@172.16.1.1
Supported: replaces
Supported: sdp-anat
....
v=0
o=CiscoSystemsSIP-GW-UserA
s=SIP Call
t=0 0
m=audio 19350 RTP/AVP 0
c=IN IP4 172.16.1.10
Which would cause inside users not to hear outside users (the other way around as described). So I am curious is there is some sort of ALG functionality in your ADSL modem perphaps. Well, cheers then.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.161.164.69:5060;branch
From: "Anonymous"<sip:anonymous@
To: "Patrick Hall"<sip:0290788356@voice
Date: Tue, 10 Jul 2012 23:48:57 GMT
Call-ID: SDtktfe01-602dd73951c091f2
CSeq: 967170713 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:0290788356@172.16.1.1
Supported: replaces
Supported: sdp-anat
....
v=0
o=CiscoSystemsSIP-GW-UserA
s=SIP Call
t=0 0
m=audio 19350 RTP/AVP 0
c=IN IP4 172.16.1.10
Which would cause inside users not to hear outside users (the other way around as described). So I am curious is there is some sort of ALG functionality in your ADSL modem perphaps. Well, cheers then.
so your adsl router is it sip enabled as that should have been stated in you question if its not you shouldn't have to configure open ports on a sip adsl router as sip routers know about the nat firewall
ASKER
I spend a lot of time to check the configuration of UC520 while the problem was on the ADSL router.