[asterisk-users] port 5060 is blocked by ISP
SamyGo
govoiper at gmail.com
Wed Jul 4 04:40:45 CDT 2012
Hi,
Being audible sometime or bad voice quality is only due to internet latency
or bad internet situation.
[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt:
Retransmission timeout reached on transmission
551b9e744a6b41ee2c00033e22b333c0 at 122.160.154.189:5060 for seqno 102
(Critical Request) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up
call 551b9e744a6b41ee2c00033e22b333c0 at 122.160.154.189:5060 - no reply to
our critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
The above lines again telling that there is some problem sending sequential
packets to some endpoint. That may lead to disconnection of call after some
time..as it is currently doing so.
Try setting some more NAT parameters...as you said localnet. Set
*localnet=*parameter entries in your asterisk server sip
configurations.
BR
Sammy
On Wed, Jul 4, 2012 at 2:13 PM, alok srivastava <alokkic at gmail.com> wrote:
> thanks Samy
> i have set nat=yes, now getting sound from both side but there is too uch
> disturbance. soetime we becoe audible and sometime not.i did not set extern
> ip coz my asterisk server is directly configured on public ip. I have
> softphones on some where localnets separate from asterisk server campus . i
> also set "sip set debug on" CLI prompt. this is giving following error.
>
> when i test sip traffic on wireshark "401 unauthorize" error getting this
> error cli prompt also showing.
>
> my first softph(9001) is on localnet 192.168.1.136 and 2nd softphone
> (9000) in another localnet in another campus(192.168.6.25)
>
>
> Scheduling destruction of SIP dialog '
> 551b9e744a6b41ee2c00033e22b333c0 at 122.160.154.189:5060' in 32000 ms
> (Method: INVITE)
> [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt:
> Retransmission timeout reached on transmission
> 551b9e744a6b41ee2c00033e22b333c0 at 122.160.154.189:5060 for seqno 102
> (Critical Request) -- See
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 32000ms with no response
> [Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up
> call 551b9e744a6b41ee2c00033e22b333c0 at 122.160.154.189:5060 - no reply to
> our critical packet (see
> https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
> -- SIP/9000-00000005 is circuit-busy
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Auto fallthrough, channel 'SIP/9001-00000004' status is 'CONGESTION'
>
> <--- Reliably Transmitting (NAT) to 122.163.193.94:1801 --->
> SIP/2.0 503 Service Unavailable
> Via: SIP/2.0/UDP 192.168.1.136:5060
> ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;received=122.163.193.94;rport=1801
> From: "9001"<sip:9001 at 122.160.154.189>;tag=b0785362
> To: <sip:9000 at 122.160.154.189>;tag=as6c7d28d1
> Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.
> CSeq: 2 INVITE
> Server: Asterisk PBX 10.0.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> X-Asterisk-HangupCause: No user responding
> X-Asterisk-HangupCauseCode: 18
> Content-Length: 0
>
>
> <------------>
> Really destroying SIP dialog '
> 551b9e744a6b41ee2c00033e22b333c0 at 122.160.154.189:5060' Method: INVITE
>
> <--- SIP read from UDP:122.163.193.94:1801 --->
> ACK sip:9000 at 122.160.154.189 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.136:5060
> ;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;rport
> Max-Forwards: 70
> To: <sip:9000 at 122.160.154.189>;tag=as6c7d28d1
> From: "9001"<sip:9001 at 122.160.154.189>;tag=b0785362
> Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.
> CSeq: 2 ACK
> Content-Length: 0
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> 'MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.' Method: ACK
>
> <--- SIP read from UDP:122.163.193.94:1801 --->
>
>
> <------------->
>
> <--- SIP read from UDP:115.249.67.250:5060 --->
> REGISTER sip:122.160.154.189 SIP/2.0
> Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKskzxkdlp
> Max-Forwards: 70
> To: "shekhar" <sip:9000 at 122.160.154.189>
> From: "shekhar" <sip:9000 at 122.160.154.189>;tag=jcysf
> Call-ID: qajoimvsihxpsff at alok-Inspiron-N5110
> CSeq: 954 REGISTER
> Contact: <sip:9000 at 192.168.6.25>;expires=3600
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> User-Agent: Twinkle/1.4.2
> Content-Length: 0
>
> <------------->
> --- (11 headers 0 lines) ---
> Sending to 115.249.67.250:5060 (NAT)
>
> <--- Transmitting (NAT) to 115.249.67.250:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 192.168.6.25;branch=z9hG4bKskzxkdlp;received=115.249.67.250;rport=5060
> From: "shekhar" <sip:9000 at 122.160.154.189>;tag=jcysf
> To: "shekhar" <sip:9000 at 122.160.154.189>;tag=as26d4cd86
> Call-ID: qajoimvsihxpsff at alok-Inspiron-N5110
> CSeq: 954 REGISTER
> Server: Asterisk PBX 10.0.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="278a3764"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'qajoimvsihxpsff at alok-Inspiron-N5110'
> in 32000 ms (Method: REGISTER)
>
> <--- SIP read from UDP:115.249.67.250:5060 --->
> REGISTER sip:122.160.154.189 SIP/2.0
> Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKjmqlllxn
> Max-Forwards: 70
> To: "shekhar" <sip:9000 at 122.160.154.189>
> From: "shekhar" <sip:9000 at 122.160.154.189>;tag=jcysf
> Call-ID: qajoimvsihxpsff at alok-Inspiron-N5110
> CSeq: 955 REGISTER
> Contact: <sip:9000 at 192.168.6.25>;expires=3600
> Authorization: Digest
> username="9000",realm="asterisk",nonce="278a3764",uri="sip:122.160.154.189",response="c7a119185514202d5f9cc10a86a93607",algorithm=MD5
> Allow:
> INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
> User-Agent: Twinkle/1.4.2
> Content-Length: 0
>
> <------------->
> --- (12 headers 0 lines) ---
> Sending to 115.249.67.250:5060 (NAT)
>
> <--- Transmitting (NAT) to 115.249.67.250:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.6.25;branch=z9hG4bKjmqlllxn;received=115.249.67.250;rport=5060
> From: "shekhar" <sip:9000 at 122.160.154.189>;tag=jcysf
> To: "shekhar" <sip:9000 at 122.160.154.189>;tag=as26d4cd86
> Call-ID: qajoimvsihxpsff at alok-Inspiron-N5110
> CSeq: 955 REGISTER
> Server: Asterisk PBX 10.0.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
> PUBLISH
> Supported: replaces, timer
> Expires: 3600
> Contact: <sip:9000 at 192.168.6.25>;expires=3600
> Date: Wed, 04 Jul 2012 14:08:17 GMT
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'qajoimvsihxpsff at alok-Inspiron-N5110'
> in 32000 ms (Method: REGISTER)
>
> <--- SIP read from UDP:122.163.193.94:1801 --->
>
>
> <------------->
> Really destroying SIP dialog 'qajoimvsihxpsff at alok-Inspiron-N5110'
> Method: REGISTER
>
>
> regards
> abhi
>
>
>
>
>
>
>
>
>
>
> On Mon, Jul 2, 2012 at 5:22 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> actually its a one-way audio issue due to NAT !
>>
>> alok , please explain your network flow for end to end
>> client-server-client.
>>
>> You may need to set nat=yes for your sip peer behind NAT. If the server
>> is behind NAT router/firewall use externip=<public.ip.of.server> field.
>> Also provide sip traces of this call.
>> Another thing to do for your learning. Execute wireshark on both
>> softphone systems and set "sip | rtp" as filter and see where are the RTP
>> streams going on each end !
>>
>> Take a complete capture on Asterisk server by executing the command "sip
>> set debug on" and make a call.
>>
>> BR
>> Sammy
>>
>>
>> On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon <digium at sanguinarius.co.uk>wrote:
>>
>>> alok srivastava wrote:
>>>
>>>> dear
>>>> i have configured properly asterisk. At the one end i am using x-lite
>>>> soft ph and another end twinkle. call is going properly from both end but
>>>> after picking the phone not able to listen other one.
>>>> when i checked the port 5060 on the asterisk server it is always
>>>> showing closed while i have flushed all the rules from iptables (iptables
>>>> -F)
>>>>
>>>> PORT STATE SERVICE VERSION
>>>> 5060/tcp closed sip
>>>>
>>>> telnet localhost 5060 (could not connect)
>>>>
>>>> regards
>>>> alok
>>>>
>>>>
>>>> SIP is only used to setup (and stop etc.) the call. The actual audio
>>> is sent via rtp.
>>>
>>> /etc/asterisk/rtp.conf
>>>
>>> Should tell which ports asterisk is using for rtp, you will need to make
>>> sure that the remote host can connect to these ports.
>>>
>>> There are lots of articles around on how to resolve this.
>>>
>>>
>>>
>>>
>>> --
>>> ______________________________**______________________________**
>>> _________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
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>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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