with wireshark i need decrypt traffic every call which is time
consuming. get debug from pjnat through asterisk is not possible because
of technical reasons or nobody did it?
in my case its strange that ice candidates are the same
good call
v=0
o=- 3669976329745317845 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo
m=audio 52421 RTP/SAVPF 8 0 101
c=IN IP4 10.2.152.36
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host
generation 0 network-id 1 network-cost 10
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype
active generation 0 network-id 1 network-cost 10
bad call
v=0
o=- 2602173234285924157 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS aDrO7zRNTqNWKodpSG62Co1IDoHReEpT8Ga3
m=audio 63249 RTP/SAVPF 8 0 101
c=IN IP4 10.2.152.36
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 63249 typ host
generation 0 network-id 1 network-cost 10
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host tcptype
active generation 0 network-id 1 network-cost 10
but RTP looks like
bad call (1.1.1.1 is "public" ip of PSTN SIP GW)
Got RTP packet from 1.1.1.1:13460 (type 08, seq 002433, ts 000160,
len 000160)
Sent RTP packet to 10.2.152.36:63249 (type 08, seq 022470, ts
000160, len 000160)
Got RTP packet from 1.1.1.1:13460 (type 08, seq 002434, ts 000320,
len 000160)
Sent RTP packet to 10.2.152.36:63249 (type 08, seq 022471, ts
000320, len 000160)
Got RTP packet from 1.1.1.1:13460 (type 08, seq 002435, ts 000480,
len 000160)
good call (1.1.1.1 is "public" ip of PSTN SIP GW, 2.2.2.2 is public IP
of router)
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021197, ts 000160,
len 000160)
Sent RTP packet to *10.2.152.36:52421 (type 08, seq 032328, ts 000160,
len 000160)*
[Dec 11 16:59:53] DEBUG[44360]: res_rtp_asterisk.c:6049
ast_rtp_remote_address_set: Setting RTCP address on RTP instance
'0x7faa14005408'
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021198, ts 000320,
len 000160)
Sent RTP packet to 2.2.2.2:52421 (via ICE) (type 08, seq 032329, ts
000320, len 000160)
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021199, ts 000480,
len 000160)
Sent RTP packet to 2.2.2.2:52421 (via ICE) (type 08, seq 032330, ts
000480, len 000160)
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021200, ts 000640,
len 000160)
Sent RTP packet to 2.2.2.2:52421 (via ICE) (type 08, seq 032331, ts
000640, len 000160)
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021201, ts 000800,
len 000160)
looking for the part where RTP engine switch from *10.2.152.36 to **2.2.2.2*
it looks like**its somewhere in the learning phase*
*
Dne 12/12/2019 v 11:51 Joshua C. Colp napsal(a):
On Thu, Dec 12, 2019 at 6:39 AM marek <cervaj...@gmail.com
<mailto:cervaj...@gmail.com>> wrote:
hi,
i have following topology
PSTN - Asterisk ---- internet ----- router - jssip client (wss)
Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp
for SIP
connection to PSTN
router - public IP/private IP (NAT)
jssip client - private IP - sip over websocket to Asterisk PJSIP
~30% of calls has problem with no audio. reason is that Asterisk is
sending RTP to private IP of jssip
SDP looks the same for good call and bad call too
i searched through res_rtp_asterisk.c but i'm not sure where to put
DEBUG info about which IP and why Asterisk pick for RTP
any hint?
is it possible debug Asterisk STUN request/response ? or is it
hidden in
pjsip internals?
ICE is performed using pjnath, which is part of pjproject and not
Asterisk itself. Looking at the SDP and the ICE candidates can tell
you the possibilities for the paths, and a wireshark capture can show
you the actual traffic going back and forth and what is being attempted.
--
Joshua C. Colp
Senior Software Developer
Sangoma Technologies
Check us out at www.sangoma.com <http://www.sangoma.com> and
www.asterisk.org <http://www.asterisk.org>
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