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<p>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?</p>
<p><br>
</p>
<p>in my case its strange that ice candidates are the same</p>
<p>good call<br>
</p>
<p>v=0<br>
o=- 3669976329745317845 2 IN IP4 127.0.0.1<br>
s=-<br>
t=0 0<br>
a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo<br>
m=audio 52421 RTP/SAVPF 8 0 101<br>
c=IN IP4 10.2.152.36<br>
a=rtcp:9 IN IP4 0.0.0.0<br>
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 52421 typ host
generation 0 network-id 1 network-cost 10<br>
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host
tcptype active generation 0 network-id 1 network-cost 10<br>
</p>
<p>bad call<br>
</p>
<p>v=0<br>
o=- 2602173234285924157 2 IN IP4 127.0.0.1<br>
s=-<br>
t=0 0<br>
a=msid-semantic: WMS aDrO7zRNTqNWKodpSG62Co1IDoHReEpT8Ga3<br>
m=audio 63249 RTP/SAVPF 8 0 101<br>
c=IN IP4 10.2.152.36<br>
a=rtcp:9 IN IP4 0.0.0.0<br>
a=candidate:3607370648 1 udp 2122260223 10.2.152.36 63249 typ host
generation 0 network-id 1 network-cost 10<br>
a=candidate:2575820648 1 tcp 1518280447 10.2.152.36 9 typ host
tcptype active generation 0 network-id 1 network-cost 10<br>
</p>
<p><br>
</p>
<p>but RTP looks like</p>
<p>bad call (1.1.1.1 is "public" ip of PSTN SIP GW)<br>
</p>
<p>Got RTP packet from 1.1.1.1:13460 (type 08, seq 002433, ts
000160, len 000160)<br>
Sent RTP packet to 10.2.152.36:63249 (type 08, seq 022470, ts
000160, len 000160)<br>
Got RTP packet from 1.1.1.1:13460 (type 08, seq 002434, ts
000320, len 000160)<br>
Sent RTP packet to 10.2.152.36:63249 (type 08, seq 022471, ts
000320, len 000160)<br>
Got RTP packet from 1.1.1.1:13460 (type 08, seq 002435, ts
000480, len 000160)<br>
</p>
<p>good call (1.1.1.1 is "public" ip of PSTN SIP GW, 2.2.2.2 is
public IP of router)</p>
<p>Got RTP packet from 1.1.1.1:15026 (type 08, seq 021197, ts
000160, len 000160)<br>
Sent RTP packet to <b>10.2.152.36:52421 (type 08, seq
032328, ts 000160, len 000160)</b></p>
<p>[Dec 11 16:59:53] DEBUG[44360]: res_rtp_asterisk.c:6049
ast_rtp_remote_address_set: Setting RTCP address on RTP instance
'0x7faa14005408'</p>
<p>Got RTP packet from 1.1.1.1:15026 (type 08, seq 021198, ts
000320, len 000160)<br>
Sent RTP packet to 2.2.2.2:52421 (via ICE) (type 08, seq
032329, ts 000320, len 000160)<br>
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021199, ts
000480, len 000160)<br>
Sent RTP packet to 2.2.2.2:52421 (via ICE) (type 08, seq
032330, ts 000480, len 000160)<br>
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021200, ts
000640, len 000160)<br>
Sent RTP packet to 2.2.2.2:52421 (via ICE) (type 08, seq
032331, ts 000640, len 000160)<br>
Got RTP packet from 1.1.1.1:15026 (type 08, seq 021201, ts
000800, len 000160)<br>
<br>
</p>
<p>looking for the part where RTP engine switch from <b>10.2.152.36
to </b><b>2.2.2.2</b></p>
<p>it looks like<b> </b>its somewhere in the learning phase<b><br>
</b></p>
<p><br>
</p>
<div class="moz-cite-prefix">Dne 12/12/2019 v 11:51 Joshua C. Colp
napsal(a):<br>
</div>
<blockquote type="cite"
cite="mid:CAM0A2Z3kAy9U1syUmK+T6EaMM7k7pmL7wJs+Y2rV34_Ydf2jXg@mail.gmail.com">
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<div dir="ltr">
<div dir="ltr">On Thu, Dec 12, 2019 at 6:39 AM marek <<a
href="mailto:cervajs64@gmail.com" moz-do-not-send="true">cervajs64@gmail.com</a>>
wrote:<br>
</div>
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left:1px solid
rgb(204,204,204);padding-left:1ex">hi,<br>
<br>
i have following topology<br>
<br>
PSTN - Asterisk ---- internet ----- router - jssip client
(wss)<br>
<br>
Asterisk 13.29.1 on public IP, chan_pjsip for wss,
chan_sip/udp for SIP <br>
connection to PSTN<br>
<br>
router - public IP/private IP (NAT)<br>
<br>
jssip client - private IP - sip over websocket to Asterisk
PJSIP<br>
<br>
<br>
~30% of calls has problem with no audio. reason is that
Asterisk is <br>
sending RTP to private IP of jssip<br>
<br>
SDP looks the same for good call and bad call too<br>
<br>
<br>
i searched through res_rtp_asterisk.c but i'm not sure where
to put <br>
DEBUG info about which IP and why Asterisk pick for RTP<br>
<br>
any hint?<br>
<br>
<br>
is it possible debug Asterisk STUN request/response ? or is
it hidden in <br>
pjsip internals?<br>
</blockquote>
<div><br>
</div>
<div>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.</div>
</div>
<div><br>
</div>
-- <br>
<div dir="ltr" class="gmail_signature">
<div dir="ltr">
<div>
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div dir="ltr">
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Joshua C. Colp</font></div>
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Senior Software Developer</font></div>
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Sangoma Technologies</font></div>
<div style="font-family:tahoma,sans-serif"><font
color="#073763">Check us out at <a
href="http://www.sangoma.com"
target="_blank" moz-do-not-send="true">www.sangoma.com</a>
and <a href="http://www.asterisk.org"
target="_blank" moz-do-not-send="true">www.asterisk.org</a></font><br>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
</div>
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