Почему пакеты RTP уходят на левый адрес?

Question

[Yermek]

Всем здравия! Настраиваю астериск через pjsip

При такой конфигурации:

pjsip.conf

[udp-transport]
type=transport
protocol=udp
bind=0.0.0.0:5699
local_net=192.168.200.0/24
local_net=127.0.0.1/32
external_media_address=55.176.150.171
external_signaling_address=55.176.150.171
external_signaling_port=5699
symmetric_transport=yes

[SP1]
type=endpoint
transport=udp-transport
context=demo
;nat begin
;rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
;nat end
disallow=all
allow=ulaw
aors=SP1
auth=authSP1
direct_media=yes

[SP1]
type=aor
max_contacts=1

[authSP1]
type=auth
auth_type=userpass
password=pass4tests
username=SP1

Имею вот такую вот картину


Проверял на demo контексте звонком на 600 (эхо тест). Причем голос от софт-фона на сервер передается (анализ RTP пакетов слушал)

Скажите пожалуйста, куда мне копать

Answer 1

[Valentin Barbolin]

direct_media=no

Полагаю, что слева клиент, его внешний адрес 93.185.73.55. Адрес 192.168.20.251, это внутренний адрес клиента.
Покажи INVITE от 93.185.73.55.

Comments

[Yermek]

UDP поток

INVITE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 192.168.20.251:5699;rport;branch=z9hG4bKPj7e7c5100-2d04-4808-be93-4d39bee2b2f4
Max-Forwards: 70
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: sip:[e-mail]
Contact: <sip:SP1@93.185.73.55:5699;ob>;+sip.ice
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
CSeq: 21647 INVITE
Route: <sip:mydomain.kz:5699;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Simotel - Android 9
Content-Type: application/sdp
Content-Length:   742

v=0
o=- 3826160558 3826160558 IN IP4 192.168.20.251
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4004 RTP/AVP 98 97 99 104 3 0 8 9 18 96
c=IN IP4 192.168.20.251
b=TIAS:64000
a=rtcp:4024 IN IP4 192.168.20.251
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ssrc:546953231 cname:699aa2bd538dd60f
a=ice-ufrag:34eacb43
a=ice-pwd:60194ea878ffa5c83ea8a0a9
a=candidate:Hc0a814fb 1 UDP 2130706431 192.168.20.251 4004 typ host
a=candidate:Hc0a814fb 2 UDP 2130706430 192.168.20.251 4024 typ host
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.251:5699;rport=5699;received=93.185.73.55;branch=z9hG4bKPj7e7c5100-2d04-4808-be93-4d39bee2b2f4
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: <sip:[e-mail]>;tag=z9hG4bKPj7e7c5100-2d04-4808-be93-4d39bee2b2f4
CSeq: 21647 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1617171759/bb318de1cbfed38c1e150032b4231a7f",opaque="046312ff651e54d5",algorithm=md5,qop="auth"
Server: Asterisk PBX certified/16.8-cert7
Content-Length:  0

ACK sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 192.168.20.251:5699;rport;branch=z9hG4bKPj7e7c5100-2d04-4808-be93-4d39bee2b2f4
Max-Forwards: 70
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: sip:[e-mail];tag=z9hG4bKPj7e7c5100-2d04-4808-be93-4d39bee2b2f4
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
CSeq: 21647 ACK
Route: <sip:mydomain.kz:5699;lr>
Content-Length:  0

INVITE sip:[e-mail] SIP/2.0
Via: SIP/2.0/UDP 93.185.73.55:5699;rport;branch=z9hG4bKPja4c0672a-4c47-4ac0-ac37-350ec0051b66
Max-Forwards: 70
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: sip:[e-mail]
Contact: <sip:SP1@93.185.73.55:5699;ob>;+sip.ice
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
CSeq: 21648 INVITE
Route: <sip:mydomain.kz:5699;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Simotel - Android 9
Authorization: Digest username="SP1", realm="asterisk", nonce="1617171759/bb318de1cbfed38c1e150032b4231a7f", uri="sip:[e-mail]", response="3d71a0def323ac74c98387097c2cac98", algorithm=md5, cnonce="378e10f2ff1d4915b9339221ed5877a4", opaque="046312ff651e54d5", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   742

v=0
o=- 3826160558 3826160558 IN IP4 192.168.20.251
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4004 RTP/AVP 98 97 99 104 3 0 8 9 18 96
c=IN IP4 192.168.20.251
b=TIAS:64000
a=rtcp:4024 IN IP4 192.168.20.251
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ssrc:546953231 cname:699aa2bd538dd60f
a=ice-ufrag:34eacb43
a=ice-pwd:60194ea878ffa5c83ea8a0a9
a=candidate:Hc0a814fb 1 UDP 2130706431 192.168.20.251 4004 typ host
a=candidate:Hc0a814fb 2 UDP 2130706430 192.168.20.251 4024 typ host
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 93.185.73.55:5699;rport=5699;received=93.185.73.55;branch=z9hG4bKPja4c0672a-4c47-4ac0-ac37-350ec0051b66
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: <sip:[e-mail]>
CSeq: 21648 INVITE
Server: Asterisk PBX certified/16.8-cert7
Content-Length:  0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.185.73.55:5699;rport=5699;received=93.185.73.55;branch=z9hG4bKPja4c0672a-4c47-4ac0-ac37-350ec0051b66
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: <sip:[e-mail]>;tag=8a316ea1-b69a-48dd-82bc-6809d68692da
CSeq: 21648 INVITE
Server: Asterisk PBX certified/16.8-cert7
Contact: <sip:55.176.150.171:5699>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   232

v=0
o=- 3826160558 3826160560 IN IP4 55.176.150.171
s=Asterisk
c=IN IP4 55.176.150.171
t=0 0
m=audio 17070 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
ACK sip:55.176.150.171:5699 SIP/2.0
Via: SIP/2.0/UDP 93.185.73.55:5699;rport;branch=z9hG4bKPjb0de839f-a88e-4640-95f5-9cd03bfd2e52
Max-Forwards: 70
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: sip:[e-mail];tag=8a316ea1-b69a-48dd-82bc-6809d68692da
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
CSeq: 21648 ACK
Content-Length:  0


BYE sip:55.176.150.171:5699 SIP/2.0
Via: SIP/2.0/UDP 93.185.73.55:5699;rport;branch=z9hG4bKPj737c04b8-ae1c-42f5-b9d3-558ad144bf53
Max-Forwards: 70
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: sip:[e-mail];tag=8a316ea1-b69a-48dd-82bc-6809d68692da
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
CSeq: 21649 BYE
User-Agent: Simotel - Android 9
Content-Length:  0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 93.185.73.55:5699;rport=5699;received=93.185.73.55;branch=z9hG4bKPj737c04b8-ae1c-42f5-b9d3-558ad144bf53
Call-ID: 97a04e74-2a9f-4c8b-abab-4b8ad085f49b
From: <sip:[e-mail]>;tag=1fbb7758-1e67-43ad-8840-79fff26aa1c1
To: <sip:[e-mail]>;tag=8a316ea1-b69a-48dd-82bc-6809d68692da
CSeq: 21649 BYE
Server: Asterisk PBX certified/16.8-cert7
Content-Length:  0

Откуда 192.168.20.251 я вообще не понимаю :(

Правильно ли я считаю, что его присылает сам сфот-фон?

[Valentin Barbolin]

Yermek, Это может WIFI адрес или мобильного интернета, может VPN поднят, не суть. Проблема в не корректной конфигурации NAT на стороне сервера. Клиент за NAT соответственно сервер должен отправлять RTP на белый адрес с которого пришел INVITE, а он шлет на адрес SDP (o=- 3826160558 3826160558 IN IP4 192.168.20.251).

https://wiki.asterisk.org/wiki/display/AST/Migrati...
[mypeer1]
type=endpoint
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

[Yermek]

Вот блин :) У меня rtp_symmetric=yes закоменчен был :) Раскомментил, и голос пошел.

Между двумя абонентами тоже пошла связь. Хотя RTP все равно на непонятный адрес уходит. Но буду это курить дальше. База самое главное есть :)