RTP трафик

Цепочка : Астериск - NAT (Mikrotik) - интернет - NAT - VoIP телефон

На микротике порты 5160 и 10000-20000 проброшены, service port SIP - отключен. Но РТП трафик в сторону VoIP телефона не идет.

Wireshark показал, что астер шлет rtp пакеты на локальный ИП VoIP телефона.

На Астериске externip и localnet указаны. Почему Астериск решает вдруг слать пакеты на локальный ип телефона?

<--- SIP read from UDP:94.BBB.CCC.DDD:1124 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 87.BBB.CCC.DDD:5160;branch=z9hG4bK4efc6710 From: "131" <sip:131@87.BBB.CCC.DDD:5160>;tag=as17b36e98 To: <sip:129@94.BBB.CCC.DDD:1124>;tag=1454414836 Call-ID: 602b9b931429ebec66556ad02233008d@87.BBB.CCC.DDD:5160 CSeq: 102 INVITE Contact: <sip:129@94.BBB.CCC.DDD:1124> Supported: 100rel, replaces, timer User-Agent: Fanvil X3S/R2.2.0.3696_0c383e1a63db Allow-Events: talk,hold Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE Content-Type: application/sdp Content-Length: 241 v=0 o=129 1860417920 1951922284 IN IP4 192.168.1.12 s=A conversation c=IN IP4 192.168.1.12 t=0 0 m=audio 10008 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) > 0x6041f300 -- Strict RTP learning after remote address set to: 192.168.1.12:10008 **Peer audio RTP is at port 192.168.1.12:10008** sip_route_dump: route/path hop: <sip:129@94.BBB.CCC.DDD:1124> set_destination: Parsing <sip:129@94.BBB.CCC.DDD:1124> for address/port to send to set_destination: set destination to 94.BBB.CCC.DDD:1124 Transmitting (no NAT) to 94.BBB.CCC.DDD:1124: ACK sip:129@94.BBB.CCC.DDD:1124 SIP/2.0 Via: SIP/2.0/UDP 87.BBB.CCC.DDD:5160;branch=z9hG4bK26df7106 Max-Forwards: 70 From: "131" <sip:131@87.BBB.CCC.DDD:5160>;tag=as17b36e98 To: <sip:129@94.BBB.CCC.DDD:1124>;tag=1454414836 Contact: <sip:131@87.BBB.CCC.DDD:5160> Call-ID: 602b9b931429ebec66556ad02233008d@87.BBB.CCC.DDD:5160 CSeq: 102 ACK User-Agent: FPBX-14.0.3.2(13.19.0) Content-Length: 0