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Увеличивается задержка (Peer is Lagged) после звонка.

https://forum.asterisk.ru/viewtopic.php?t=1235

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Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 09 янв 2012, 16:39
drTr0jan
Заметил одну нехорошую проблему во взаимодействии с софтсвичем оператора.
Звоню через астер и софтсвич оператора на город. После завершения разговора, если я кладу трубку, то временно увеличивается задержка пира:
[Jan 9 23:20:53] NOTICE[60372]: chan_sip.c:20249 handle_response_peerpoke: Peer 'ttk' is now Lagged. (43468ms / 2000ms)
[Jan 9 23:21:03] NOTICE[60372]: chan_sip.c:20249 handle_response_peerpoke: Peer 'ttk' is now Reachable. (9ms / 2000ms)
Собственно, в этот промежуток времени невозможно совершать звонки через этот пир.

Если же после завершения разговора кладёт трубку вызываемый абонент, то такой проблемы не наблюдается.

Можно конечно отключить qualify, но это предупреждение всё равно будет сыпаться в логи. Из-за чего такое происходит? Куда копать?

Нормальный звонок:

Код: Выделить всё

SIP Debugging Enabled for IP: 172.31.1.34
    -- Executing [420000@office:1] Dial("SIP/201-0000002f", "SIP/ttk2/420000,,T") in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
INVITE sip:420000@172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK48b96f01
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as147152a9
To: <sip:420000@172.31.1.34>
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 26 Dec 2011 15:14:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1838601133 1838601133 IN IP4 172.30.1.206
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.30.1.206
t=0 0
m=audio 19492 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/ttk2/420000

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK48b96f01
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
To: <sip:420000@172.31.1.34>
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK48b96f01
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
To: <sip:420000@172.31.1.34>;tag=8ec03f1e
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:420000@172.31.1.34:5060;user=phone>
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9557516 9557516 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 30912 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:30912
    -- SIP/ttk2-00000030 is ringing
    -- SIP/ttk2-00000030 is making progress passing it to SIP/201-0000002f

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK48b96f01
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
To: <sip:420000@172.31.1.34>;tag=8ec03f1e
CSeq: 102 INVITE
Contact: <sip:420000@172.31.1.34:5060;user=phone>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9557516 9557517 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 30912 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:30912
list_route: hop: <sip:420000@172.31.1.34:5060;user=phone>
set_destination: Parsing <sip:420000@172.31.1.34:5060;user=phone> for address/port to send to
set_destination: set destination to 172.31.1.34:5060
Transmitting (no NAT) to 172.31.1.34:5060:
ACK sip:420000@172.31.1.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK14ec0f9f
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as147152a9
To: <sip:420000@172.31.1.34>;tag=8ec03f1e
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


---
    -- SIP/ttk2-00000030 answered SIP/201-0000002f

<--- SIP read from UDP:172.31.1.34:5060 --->
OPTIONS sip:910000@172.30.1.206:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bK6b45c7e8928ac7b10ed53992f
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
From: <sip:420000@172.31.1.34>;tag=8ec03f1e
To: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
CSeq: 1 OPTIONS
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- Transmitting (no NAT) to 172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bK6b45c7e8928ac7b10ed53992f;received=172.31.1.34
From: <sip:420000@172.31.1.34>;tag=8ec03f1e
To: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:910000@172.30.1.206:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
OPTIONS sip:172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK4a0448bf
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.30.1.206>;tag=as03924cb3
To: <sip:172.31.1.34>
Contact: <sip:Unknown@172.30.1.206:5060>
Call-ID: 3b64dce84982a22e076f41037860b0c8@172.30.1.206:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 26 Dec 2011 15:14:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK4a0448bf
Call-ID: 3b64dce84982a22e076f41037860b0c8@172.30.1.206:5060
From: "Unknown"<sip:Unknown@172.30.1.206>;tag=as03924cb3
To: <sip:172.16.122.4;user=phone>;tag=a4a06d76
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3b64dce84982a22e076f41037860b0c8@172.30.1.206:5060' Method: OPTIONS

<--- SIP read from UDP:172.31.1.34:5060 --->
BYE sip:910000@172.30.1.206:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bKd8d4a10e723dec10ee633d7f4
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
From: <sip:420000@172.31.1.34>;tag=8ec03f1e
To: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
CSeq: 2 BYE
Max-Forwards: 70
Reason: Q.850;cause=16;text="normal call clearing"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 172.31.1.34:5060 (no NAT)
Scheduling destruction of SIP dialog '0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bKd8d4a10e723dec10ee633d7f4;received=172.31.1.34
From: <sip:420000@172.31.1.34>;tag=8ec03f1e
To: "Test2"<sip:910000@172.30.1.206>;tag=as147152a9
Call-ID: 0144eab0359f76f80fa4d484381b3ac5@172.30.1.206:5060
CSeq: 2 BYE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (office, 420000, 1) exited non-zero on 'SIP/201-0000002f'
tracer-tong*CLI> sip set debug off
SIP Debugging Disabled
Проблемный звонок:

Код: Выделить всё

SIP Debugging Enabled for IP: 172.31.1.34
    -- Executing [322322@office:1] Dial("SIP/201-0000000e", "SIP/ttk2/322322,,T") in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
INVITE sip:322322@172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK3ef13ca5
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 26 Dec 2011 13:48:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 2048620859 2048620859 IN IP4 172.30.1.206
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.30.1.206
t=0 0
m=audio 15706 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/ttk2/322322

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK3ef13ca5
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK3ef13ca5
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>;tag=6e729246
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:322322@172.31.1.34:5060;user=phone>
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9557146 9557146 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 30736 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:30736
    -- SIP/ttk2-0000000f is making progress passing it to SIP/201-0000000e

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK3ef13ca5
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>;tag=6e729246
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:322322@172.31.1.34:5060;user=phone>
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
    -- SIP/ttk2-0000000f is ringing

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK3ef13ca5
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>;tag=6e729246
CSeq: 102 INVITE
Contact: <sip:322322@172.31.1.34:5060;user=phone>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9557146 9557147 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 30736 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:30736
list_route: hop: <sip:322322@172.31.1.34:5060;user=phone>
set_destination: Parsing <sip:322322@172.31.1.34:5060;user=phone> for address/port to send to
set_destination: set destination to 172.31.1.34:5060
Transmitting (no NAT) to 172.31.1.34:5060:
ACK sip:322322@172.31.1.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK1b35524a
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>;tag=6e729246
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


---
    -- SIP/ttk2-0000000f answered SIP/201-0000000e

<--- SIP read from UDP:172.31.1.34:5060 --->
OPTIONS sip:910000@172.30.1.206:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bKe5360720dd30c8f48f88a8627
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
From: <sip:322322@172.31.1.34>;tag=6e729246
To: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
CSeq: 1 OPTIONS
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- Transmitting (no NAT) to 172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bKe5360720dd30c8f48f88a8627;received=172.31.1.34
From: <sip:322322@172.31.1.34>;tag=6e729246
To: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:910000@172.30.1.206:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
OPTIONS sip:172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK2b45fa2b
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.30.1.206>;tag=as0e871056
To: <sip:172.31.1.34>
Contact: <sip:Unknown@172.30.1.206:5060>
Call-ID: 60e42755545d5f0a4fb9619e49c6aa5d@172.30.1.206:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 26 Dec 2011 13:48:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK2b45fa2b
Call-ID: 60e42755545d5f0a4fb9619e49c6aa5d@172.30.1.206:5060
From: "Unknown"<sip:Unknown@172.30.1.206>;tag=as0e871056
To: <sip:172.16.122.4;user=phone>;tag=a5259fc5
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '60e42755545d5f0a4fb9619e49c6aa5d@172.30.1.206:5060' Method: OPTIONS
Scheduling destruction of SIP dialog '4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060' in 6400 ms (Method: OPTIONS)
set_destination: Parsing <sip:322322@172.31.1.34:5060;user=phone> for address/port to send to
set_destination: set destination to 172.31.1.34:5060
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
BYE sip:322322@172.31.1.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK749a9b9a
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>;tag=6e729246
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.8.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (office, 322322, 1) exited non-zero on 'SIP/201-0000000e'

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK749a9b9a
Call-ID: 4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as65eb7b9f
To: <sip:322322@172.31.1.34>;tag=6e729246
CSeq: 103 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
[Dec 27 00:48:26] NOTICE[60372]: chan_sip.c:20249 handle_response_peerpoke: Peer 'ttk2' is now Lagged. (4539ms / 2000ms)
Really destroying SIP dialog '4258239f5c36a01a4064fedf422d99f7@172.30.1.206:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
OPTIONS sip:172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK37d53ae3
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.30.1.206>;tag=as1977acd8
To: <sip:172.31.1.34>
Contact: <sip:Unknown@172.30.1.206:5060>
Call-ID: 6198434c605bded456d81e053440e4dc@172.30.1.206:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 26 Dec 2011 13:48:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:172.31.1.34:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK37d53ae3
Call-ID: 6198434c605bded456d81e053440e4dc@172.30.1.206:5060
From: "Unknown"<sip:Unknown@172.30.1.206>;tag=as1977acd8
To: <sip:172.16.122.4;user=phone>;tag=41edb563
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Encoding: identity
Accept-Language: en
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
[Dec 27 00:48:36] NOTICE[60372]: chan_sip.c:20249 handle_response_peerpoke: Peer 'ttk2' is now Reachable. (8ms / 2000ms)
Really destroying SIP dialog '6198434c605bded456d81e053440e4dc@172.30.1.206:5060' Method: OPTIONS
tracer-tong*CLI> sip set debug off
SIP Debugging Disabled

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 09 янв 2012, 18:58
zzuz
Можно конечно отключить qualify, но это предупреждение всё равно будет сыпаться в логи.
Откуда такое мнение?

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 10 янв 2012, 03:40
drTr0jan
zzuz

Из практического опыта:

Код: Выделить всё

[Jan 10 00:42:52] NOTICE[60372] chan_sip.c: Peer 'ttk' is now Lagged. (86088ms / 0ms)
[Jan 10 00:44:34] NOTICE[60372] chan_sip.c: Peer 'ttk' is now Lagged. (187757ms / 0ms)
[Jan 10 00:46:47] NOTICE[60372] chan_sip.c: Peer 'ttk' is now Lagged. (320507ms / 0ms)
Да и всё равно это не нормальная ситуация. Кстати, на другом транке (с регистрацией по логину и паролю) такой проблемы нет.

Код: Выделить всё

    -- Executing [420000@office:1] Dial("SIP/201-000001a1", "SIP/420000@ttk-old,,T") in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.124.4:5060:
INVITE sip:420000@172.16.124.4 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK0e67534f
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 09 Jan 2012 13:59:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1039137045 1039137045 IN IP4 172.30.1.206
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.30.1.206
t=0 0
m=audio 18892 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/420000@ttk-old

<--- SIP read from UDP:172.16.124.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK0e67534f;rport=5060
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.16.124.4:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK0e67534f;rport=5060
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=af7d79f7
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="Huawei",nonce="0:57:51:14857", stale=false,algorithm=MD5
Reason: Q.850;cause=0;text="unknown"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 172.16.124.4:5060:
ACK sip:420000@172.16.124.4 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK0e67534f
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=af7d79f7
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


---
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.16.124.4:5060:
INVITE sip:420000@172.16.124.4 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK61c55848
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Proxy-Authorization: Digest username="910000", realm="Huawei", algorithm=MD5, uri="sip:420000@172.16.124.4", nonce="0:57:51:14857", response="b95082d6852e4e441530d68b7e712642"
Date: Mon, 09 Jan 2012 13:59:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1039137045 1039137046 IN IP4 172.30.1.206
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.30.1.206
t=0 0
m=audio 18892 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-
<--- SIP read from UDP:172.16.124.4:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK61c55848;rport=5060
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:172.16.124.4:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK61c55848;rport=5060
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=1c5742d2
CSeq: 103 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: <sip:420000@172.16.124.4:5060;user=phone>
Content-Length: 205
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9875008 9875008 IN IP4 172.16.124.4
s=Sip Call
c=IN IP4 172.16.124.4
t=0 0
m=audio 48914 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.124.4:48914
    -- SIP/ttk-old-000001a2 is ringing
    -- SIP/ttk-old-000001a2 is making progress passing it to SIP/201-000001a1

<--- SIP read from UDP:172.16.124.4:5060 --->
hello
<------------->
Really destroying SIP dialog '3046dcd0388db4df2908655b6bbfc8ba@192.168.0.8' Method: REGISTER

<--- SIP read from UDP:172.16.124.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK61c55848;rport=5060
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=1c5742d2
CSeq: 103 INVITE
Contact: <sip:420000@172.16.124.4:5060;user=phone>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 205
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9875008 9875009 IN IP4 172.16.124.4
s=Sip Call
c=IN IP4 172.16.124.4
t=0 0
m=audio 48914 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.16.124.4:48914
list_route: hop: <sip:420000@172.16.124.4:5060;user=phone>
set_destination: Parsing <sip:420000@172.16.124.4:5060;user=phone> for address/port to send to
set_destination: set destination to 172.16.124.4:5060
Transmitting (no NAT) to 172.16.124.4:5060:
ACK sip:420000@172.16.124.4:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK478aecd2
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=1c5742d2
Contact: <sip:910000@172.30.1.206:5060>
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


---
    -- SIP/ttk-old-000001a2 answered SIP/201-000001a1
Scheduling destruction of SIP dialog '0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:420000@172.16.124.4:5060;user=phone> for address/port to send to
set_destination: set destination to 172.16.124.4:5060
Reliably Transmitting (no NAT) to 172.16.124.4:5060:
BYE sip:420000@172.16.124.4:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK096c1cce
Max-Forwards: 70
From: "Test2" <sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=1c5742d2
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.8.8.0
Proxy-Authorization: Digest username="910000", realm="Huawei", algorithm=MD5, uri="sip:420000@172.16.124.4:5060", nonce="0:57:51:14857", response="38e33e50022198cec2e22d7b723e7b6e"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (office, 420000, 1) exited non-zero on 'SIP/201-000001a1'

<--- SIP read from UDP:172.16.124.4:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK096c1cce;rport=5060
Call-ID: 0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060
From: "Test2"<sip:910000@172.30.1.206>;tag=as2ab86261
To: <sip:420000@172.16.124.4>;tag=1c5742d2
CSeq: 104 BYE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '0b5d7302459b9d084111f60f3b0b2251@172.30.1.206:5060' Method: INVITE

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 10 янв 2012, 09:50
Sfinx
Transmitting (no NAT) to 172.16.124.4:5060:
Похоже на недонастроенный NAT

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 10 янв 2012, 12:11
drTr0jan
switch, отвечает конечно. Отправляются с моего сервера. Роутера никакого нет. Сервак напрямую в сеть провайдера воткнут (ip астера 172.30.1.206).

Sfinx, никакого NATа нет.

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 11 янв 2012, 02:25
Sfinx
172.16. это натовская сетка, хоть тресни - http://ru.wikipedia.org/wiki/%D0%A7%D0% ... 0%B5%D1%81

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 11 янв 2012, 05:04
drTr0jan
Sfinx
Тресни.
Мой астер (172.30.1.206) и софтсвич оператора (172.31.1.34) находятся в одной сетке (172.16.0.0/12). Которая к интернету никаким боком не подсоединена. Причём тут NAT то?
Софтсвич оператора терминирует у себя ТфОП, на которую я звоню и имею проблему.

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 11 янв 2012, 09:42
drTr0jan
switch, своевременны. Реальный пинг прежний.

Во время простоя задержка (qualify) прыгает с 2 до 15 мс. И только после завершения разговора (при вышеописанных условиях) подрастает до 1000-60000 мс (в зависимости от длительности разговора).

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 11 янв 2012, 10:00
Vlad1983
qualifyfreq=300
или вообще
qualify=no

Re: Увеличивается задержка (Peer is Lagged) после звонка.

Добавлено: 11 янв 2012, 11:10
Sfinx
drTr0jan писал(а):Sfinx
Тресни.
Мой астер (172.30.1.206) и софтсвич оператора (172.31.1.34) находятся в одной сетке (172.16.0.0/12). Которая к интернету никаким боком не подсоединена. Причём тут NAT то?
Софтсвич оператора терминирует у себя ТфОП, на которую я звоню и имею проблему.
Ежики плакали, кололись, но продолжали не замечать NAT ...
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