Диагностика sip rtp

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GuonniBox
Сообщения: 10
Зарегистрирован: 27 апр 2017, 05:24

Диагностика sip rtp

Сообщение GuonniBox » 02 июн 2018, 15:39

Имеются Две АТС (Asterisk и Hipath 3550+HG1500). Есть проблема с вызовами. Из Asterisk в Hipath 3550 не проходит голосовой трафик, в трубке тишина. Сигнализация в обе стороны проходит, голосовой трафик из Hipath 3550 в Asterisk есть. Думаю проблема с RTP трафиком, раз SIP работает.

Asterisk = 192.168.1.17/24
Hipath 3550 = 192.168.3.17/24 (HG1500 это VoIP модуль 192.168.3.117/24)


На Asterisk нумерация = 1XXX
На Hipath 3550 нумерация = 3XXX ( внутренние номера XXX + 3, когда звонишь на внешние номера ( функция DID))

Был произведен вызов из 3101 в 1002.


На Asterisk в режиме CLI Собран sip dump.
Ниже инфа, что я собрал из хронологии sip dump, еще ниже сам sip dump.



SIP порты

От Сименса source port 5060
От Астериска source port 5060



RTP порты


Сообщение INVITE

Сименс ожидает трафик на своем порту 29100


Сообщение 100 Trying

Астериск ожидает трафик на своем порту 10886


Сообщение 183

Астериск ожидает трафик на своем порту 10886


Сообщение 200 OK

Астериск ожидает трафик на своем порту 10886


Сообщение ACK

Сименс ожидает трафик на своем порту 10068


Сообщение 200 OK

Сименс ожидает трафик на своем порту 29100


Сообщение INVITE

Сам IP телефон 192.168.1.199 ожидает на порту 10068


Сообщение 200 OK

Сименс ожидает трафик на своем порту 29100





local*CLI>

<--- SIP read from UDP:192.168.3.117:5060 --->
INVITE sip:1002@192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bK7680ff0332b9a4735.fe30ec7d9596fa537;rport
Max-Forwards: 70
From: 101 <sip:101@192.168.3.117>;tag=0baa5be264
To: <sip:1002@192.168.1.17>
Call-ID: 4aba9e3096e214fd
CSeq: 5188 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE
Contact: <sip:101@192.168.3.117:5060>
Min-SE: 90
P-Asserted-Identity: <sip:101@192.168.3.117>
Session-Expires: 1800
Supported: timer
User-Agent: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
Content-Type: application/sdp
Content-Length: 323

v=0
o=MxSIP 0 861587293 IN IP4 192.168.3.117
s=SIP Call
c=IN IP4 192.168.3.117
t=0 0
m=audio 29100 RTP/AVP 8 0 18 4 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:98 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:98 0-15
a=sendrecv
<------------->
--- (16 headers 15 lines) ---
Sending to 192.168.3.117:5060 (no NAT)
Sending to 192.168.3.117:5060 (no NAT)
Using INVITE request as basis request - 4aba9e3096e214fd
Found peer 'HG1500SIP' for '101' from 192.168.3.117:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 98
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw), peer - audio=(g723|ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.117:29100
Looking for 1002 in HiPath_3500_incoming (domain 192.168.1.17)
list_route: hop: <sip:101@192.168.3.117:5060>

<--- Transmitting (no NAT) to 192.168.3.117:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bK7680ff0332b9a4735.fe30ec7d9596fa537;received=192.168.3.117;rport=5060
From: 101 <sip:101@192.168.3.117>;tag=0baa5be264
To: <sip:1002@192.168.1.17>
Call-ID: 4aba9e3096e214fd
CSeq: 5188 INVITE
Server: Asterisk PBX 11.25.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1002@192.168.1.17:5060>
Content-Length: 0


<------------>
Audio is at 10886
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.3.117:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bK7680ff0332b9a4735.fe30ec7d9596fa537;received=192.168.3.117;rport=5060
From: 101 <sip:101@192.168.3.117>;tag=0baa5be264
To: <sip:1002@192.168.1.17>;tag=as647c4ac4
Call-ID: 4aba9e3096e214fd
CSeq: 5188 INVITE
Server: Asterisk PBX 11.25.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1002@192.168.1.17:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 255

v=0
o=root 223335524 223335524 IN IP4 192.168.1.17
s=Asterisk PBX 11.25.2
c=IN IP4 192.168.1.17
t=0 0
m=audio 10886 RTP/AVP 8 0 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=sendrecv

<------------>
[2018-06-03 03:54:51] NOTICE[1777]: chan_sip.c:15251 sip_reg_timeout: -- Registration for 'Server1@192.168.2.17' timed out, trying again (Attempt #117)
Audio is at 10886
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.3.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bK7680ff0332b9a4735.fe30ec7d9596fa537;received=192.168.3.117;rport=5060
From: 101 <sip:101@192.168.3.117>;tag=0baa5be264
To: <sip:1002@192.168.1.17>;tag=as647c4ac4
Call-ID: 4aba9e3096e214fd
CSeq: 5188 INVITE
Server: Asterisk PBX 11.25.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:1002@192.168.1.17:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 255

v=0
o=root 223335524 223335524 IN IP4 192.168.1.17
s=Asterisk PBX 11.25.2
c=IN IP4 192.168.1.17
t=0 0
m=audio 10886 RTP/AVP 8 0 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:192.168.3.117:5060 --->
ACK sip:1002@192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bKb36defd4d182465b1.43c03b1fb444b2934;rport
Max-Forwards: 70
From: 101 <sip:101@192.168.3.117>;tag=0baa5be264
To: <sip:1002@192.168.1.17>;tag=as647c4ac4
Call-ID: 4aba9e3096e214fd
CSeq: 5188 ACK
User-Agent: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
set_destination: Parsing <sip:101@192.168.3.117:5060> for address/port to send to
set_destination: set destination to 192.168.3.117:5060
Audio is at 10886
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.3.117:5060:
INVITE sip:101@192.168.3.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK201da6b1;rport
Max-Forwards: 70
From: <sip:1002@192.168.1.17>;tag=as647c4ac4
To: 101 <sip:101@192.168.3.117>;tag=0baa5be264
Contact: <sip:1002@192.168.1.17:5060>
Call-ID: 4aba9e3096e214fd
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.2
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 223335524 223335525 IN IP4 192.168.1.199
s=Asterisk PBX 11.25.2
c=IN IP4 192.168.1.199
t=0 0
m=audio 10068 RTP/AVP 8 0 98
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.3.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK201da6b1;rport=5060;received=192.168.1.17
From: <sip:1002@192.168.1.17>;tag=as647c4ac4
To: 101 <sip:101@192.168.3.117>;tag=0baa5be264
Call-ID: 4aba9e3096e214fd
CSeq: 102 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE
Contact: <sip:101@192.168.3.117:5060>
Require: timer
Server: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=MxSIP 0 861587294 IN IP4 192.168.3.117
s=SIP Call
c=IN IP4 192.168.3.117
t=0 0
m=audio 29100 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:98 0-15
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 98
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 98
Capabilities: us - (ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.117:29100
set_destination: Parsing <sip:101@192.168.3.117:5060> for address/port to send to
set_destination: set destination to 192.168.3.117:5060
Transmitting (no NAT) to 192.168.3.117:5060:
ACK sip:101@192.168.3.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK16911d1b;rport
Max-Forwards: 70
From: <sip:1002@192.168.1.17>;tag=as647c4ac4
To: 101 <sip:101@192.168.3.117>;tag=0baa5be264
Contact: <sip:1002@192.168.1.17:5060>
Call-ID: 4aba9e3096e214fd
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.2
Content-Length: 0


---
set_destination: Parsing <sip:101@192.168.3.117:5060> for address/port to send to
set_destination: set destination to 192.168.3.117:5060
Audio is at 10886
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.3.117:5060:
INVITE sip:101@192.168.3.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK7cdd1097;rport
Max-Forwards: 70
From: <sip:1002@192.168.1.17>;tag=as647c4ac4
To: 101 <sip:101@192.168.3.117>;tag=0baa5be264
Contact: <sip:1002@192.168.1.17:5060>
Call-ID: 4aba9e3096e214fd
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.25.2
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 223335524 223335526 IN IP4 192.168.1.199
s=Asterisk PBX 11.25.2
c=IN IP4 192.168.1.199
t=0 0
m=audio 10068 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.3.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK7cdd1097;rport=5060;received=192.168.1.17
From: <sip:1002@192.168.1.17>;tag=as647c4ac4
To: 101 <sip:101@192.168.3.117>;tag=0baa5be264
Call-ID: 4aba9e3096e214fd
CSeq: 103 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, INFO, PRACK, UPDATE
Contact: <sip:101@192.168.3.117:5060>
Require: timer
Server: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
Session-Expires: 1800;refresher=uac
Supported: timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=MxSIP 0 861587294 IN IP4 192.168.3.117
s=SIP Call
c=IN IP4 192.168.3.117
t=0 0
m=audio 29100 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:98 0-15
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
set_destination: Parsing <sip:101@192.168.3.117:5060> for address/port to send to
set_destination: set destination to 192.168.3.117:5060
Transmitting (no NAT) to 192.168.3.117:5060:
ACK sip:101@192.168.3.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.17:5060;branch=z9hG4bK73ca6769;rport
Max-Forwards: 70
From: <sip:1002@192.168.1.17>;tag=as647c4ac4
To: 101 <sip:101@192.168.3.117>;tag=0baa5be264
Contact: <sip:1002@192.168.1.17:5060>
Call-ID: 4aba9e3096e214fd
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.25.2
Content-Length: 0


_______________________________________________________________________________

Завершение вызова


local*CLI>
[2018-06-03 04:04:31] NOTICE[1777]: chan_sip.c:15251 sip_reg_timeout: -- Registration for 'Server1@192.168.2.17' timed out, trying again (Attempt #146)

<--- SIP read from UDP:192.168.3.117:5060 --->
BYE sip:1002@192.168.1.17:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bK9f85f5c505407153d.d5291bdeb3eb8d877;rport
Max-Forwards: 70
From: 101 <sip:101@192.168.3.117>;tag=381b798394
To: <sip:1002@192.168.1.17>;tag=as0578b64c
Call-ID: 3481a5a594fdbc18
CSeq: 5788 BYE
Reason: Q.850; cause=16; text="Normal call clearing"
Supported: timer
User-Agent: HiPath 3000 V8 M5T SIP Stack/4.0.26.26
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.3.117:5060 (no NAT)
Scheduling destruction of SIP dialog '3481a5a594fdbc18' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.3.117:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.117:5060;branch=z9hG4bK9f85f5c505407153d.d5291bdeb3eb8d877;received=192.168.3.117;rport=5060
From: 101 <sip:101@192.168.3.117>;tag=381b798394
To: <sip:1002@192.168.1.17>;tag=as0578b64c
Call-ID: 3481a5a594fdbc18
CSeq: 5788 BYE
Server: Asterisk PBX 11.25.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


_________________________________________________________________________________

Трафик до начала вызова



<------------>
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006239, ts 000160, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006240, ts 000320, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006241, ts 000480, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006242, ts 000640, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006243, ts 000800, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006244, ts 000960, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006245, ts 001120, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006246, ts 001280, len 000160)
Got RTP packet from 192.168.3.117:29100 (type 08, seq 000000, ts 000000, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006247, ts 001440, len 000160)
Got RTP packet from 192.168.3.117:29100 (type 08, seq 000001, ts 000160, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006248, ts 001600, len 000160)
Got RTP packet from 192.168.3.117:29100 (type 08, seq 000002, ts 000320, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006249, ts 001760, len 000160)
Got RTP packet from 192.168.3.117:29100 (type 08, seq 000003, ts 000480, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006250, ts 001920, len 000160)
Got RTP packet from 192.168.3.117:29100 (type 08, seq 000004, ts 000640, len 000160)
Sent RTP packet to 192.168.3.117:29100 (type 08, seq 006251, ts 002080, len 000160)

На Asterisk по умолчанию нет файла, явного задающего диапазон портов под rtp. Нашел ифну, что стандартное значение 10000 - 20000, но мое заключение, что он намного шире, раз принял запрос от Сименса 29100. С другой стороны если предположить, что на Астериске он шире, ясно становиться почему в Сименсе не слышно собеседника 1002. На Сименсе диапазон 29100 - 29131, но Астериск пихает ему 10886.

awsswa
Сообщения: 1713
Зарегистрирован: 28 апр 2012, 10:19
Откуда: Russia, Пермь

Re: Диагностика sip rtp

Сообщение awsswa » 03 июн 2018, 13:04

логичный вопрос - а зачем re-invite в сторону телефона ?
где directmedia=no ?
платный суппорт по мере возможностей

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