<div>Dear All,</div>
<div> </div>
<div>I am having some problems with connecting with a UA. Sometimes there is not sound in the call made, sometimes the caller would near no sound, while the callee can hear the caller. I have attached the rtp debug and sip debug for you comments. Please help me. Thank you all.
</div>
<div> </div>
<div> </div>
<div>Asterisk Version is 1.2.1</div>
<div>Asterisk RTP Range is 10000 to 20000</div>
<div>UA Listen RTP Port is 15000</div>
<div> </div>
<div> </div>
<div>Below is the the SIP Logs</div>
<div> </div>
<div>TestServer*CLI> <br><-- SIP read from <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>: <br>REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060
</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:74494a-1654e-43ce24ec@XXX.XXX.XX.XXX">
74494a-1654e-43ce24ec@XXX.XXX.XX.XXX</a><br>CSeq: 1 REGISTER<br>Contact: "XXXXXX" <sip:XXXXXX@172.28.174.25:5060><br>User-Agent: XXXXX UserAgent/1.0<br>Expires: 5000<br>Max-Forwards: 70<br>Content-Length: 0
</div>
<div>
<p><br>--- (11 headers 0 lines)---<br>Using latest REGISTER request as basis request<br>Sending to <a href="http://172.28.174.25">172.28.174.25</a> : 5060 (non-NAT)<br>Transmitting (NAT) to <a href="http://66.193.155.2:46478">
66.193.155.2:46478</a>:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a>;received=<a href="http://66.193.155.2">66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">
sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:74494a-1654e-43ce24ec@XXX.XXX.XX.XXX">74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
</a><br>CSeq: 1 REGISTER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX
</a>><br>Content-Length: 0</p>
<p><br>---<br>Transmitting (NAT) to <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a>;received=<a href="http://66.193.155.2">
66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=as504de7b8
<br>Call-ID: <a href="mailto:74494a-1654e-43ce24ec@XXX.XXX.XX.XXX">74494a-1654e-43ce24ec@XXX.XXX.XX.XXX</a><br>CSeq: 1 REGISTER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>WWW-Authenticate: Digest realm="asterisk", nonce="31e7aa76"<br>Content-Length: 0</p>
<p><br>---<br>Scheduling destruction of call <a href="mailto:'74494a-1654e-43ce24ec@XXX.XXX.XX.XXX'">'74494a-1654e-43ce24ec@XXX.XXX.XX.XXX'</a> in 15000 ms<br>TestServer*CLI> <br><-- SIP read from <a href="http://66.193.155.2:46478">
66.193.155.2:46478</a>: <br>REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX
</a>><br>To: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:74494a-1654e-43ce24ec@XXX.XXX.XX.XXX">74494a-1654e-43ce24ec@XXX.XXX.XX.XXX</a><br>CSeq: 2 REGISTER
<br>Contact: "XXXXXX" <sip:XXXXXX@172.28.174.25:5060><br>Authorization: Digest username="XXXXXX", realm="asterisk", nonce="31e7aa76", uri="sip:XXX.XXX.XX.XXX", response="3e54ea5e3a3b6df1e5db7b4a3182e18f", algorithm=MD5
<br>User-Agent: XXXXX UserAgent/1.0<br>Expires: 5000<br>Max-Forwards: 70<br>Content-Length: 0</p>
<p><br>--- (12 headers 0 lines)---<br>Using latest REGISTER request as basis request<br>Sending to <a href="http://172.28.174.25">172.28.174.25</a> : 5060 (NAT)<br>Transmitting (NAT) to <a href="http://66.193.155.2:46478">
66.193.155.2:46478</a>:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a>;received=<a href="http://66.193.155.2">66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">
sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:74494a-1654e-43ce24ec@XXX.XXX.XX.XXX">74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
</a><br>CSeq: 2 REGISTER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX
</a>><br>Content-Length: 0</p>
<p><br>---<br>Transmitting (NAT) to <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a>;received=<a href="http://66.193.155.2">
66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>><br>To: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=as504de7b8
<br>Call-ID: <a href="mailto:74494a-1654e-43ce24ec@XXX.XXX.XX.XXX">74494a-1654e-43ce24ec@XXX.XXX.XX.XXX</a><br>CSeq: 2 REGISTER<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Max-Forwards: 70<br>Expires: 3600<br>Contact: <sip:XXXXXX@172.28.174.25:5060>;expires=3600<br>Date: Wed, 18 Jan 2006 11:20:15 GMT<br>Content-Length: 0</p>
<p><br>---<br>Scheduling destruction of call <a href="mailto:'74494a-1654e-43ce24ec@XXX.XXX.XX.XXX'">'74494a-1654e-43ce24ec@XXX.XXX.XX.XXX'</a> in 15000 ms<br>Destroying call <a href="mailto:'4CEFE29BD3864FFFA788C2D43F4E6FBD@wawan'">
'4CEFE29BD3864FFFA788C2D43F4E6FBD@wawan'</a><br>TestServer*CLI> <br><-- SIP read from <a href="http://202.83.167.103:30928">202.83.167.103:30928</a>: </p>
<p><br>--- (0 headers 0 lines) Nat keepalive ---<br>TestServer*CLI> <br><-- SIP read from <a href="http://218.111.26.5:5060">218.111.26.5:5060</a>: </p>
<p><br>--- (0 headers 0 lines) Nat keepalive ---<br>Destroying call <a href="mailto:'IvDHn41FxnNwCePq@192.168.1.35'">'IvDHn41FxnNwCePq@192.168.1.35'</a><br>TestServer*CLI> <br><-- SIP read from <a href="http://66.193.155.2:46478">
66.193.155.2:46478</a>: <br>INVITE sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX
</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>
CSeq: 3 INVITE<br>Contact: <sip:XXXXXX@172.28.174.25:5060><br>Subject: no subject<br>User-Agent: XXXXX UserAgent/1.0<br>Max-Forwards: 70<br>Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER<br>Content-Type: application/SDP
<br>Accept: application/SDP, text/plain<br>Accept-Encoding: identity<br>Content-Length: 287</p>
<p>v=0<br>o=XXXXXX 11375833482 11375833482 IN IP4 <a href="http://172.28.174.25">172.28.174.25</a><br>s=VaxSoft Inc.<br>c=IN IP4 <a href="http://172.28.174.25">172.28.174.25</a><br>t=0 0<br>m=audio 15000 RTP/AVP 3 98 8 0 101
<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:98 iLBC/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16</p>
<p>--- (15 headers 12 lines)---<br>Using INVITE request as basis request - <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>Sending to <a href="http://172.28.174.25">172.28.174.25
</a> : 5060 (non-NAT)<br>Reliably Transmitting (NAT) to <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>:<br>SIP/2.0 407 Proxy Authentication Required<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060
</a>;received=<a href="http://66.193.155.2">66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">
sip:123456788@XXX.XXX.XX.XXX</a>>;tag=as733e9adc<br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>CSeq: 3 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>><br>Proxy-Authenticate: Digest realm="asterisk", nonce="452293c4"<br>Content-Length: 0
</p>
<p><br>---<br>Scheduling destruction of call <a href="mailto:'7469ea-26568-43ce24f4@172.28.174.25'">'7469ea-26568-43ce24f4@172.28.174.25'</a> in 15000 ms<br>Found user 'XXXXXX'<br>TestServer*CLI> <br><-- SIP read from
<a href="http://66.193.155.2:46478">66.193.155.2:46478</a>: <br>ACK sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">
sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>>;tag=as733e9adc<br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25
</a><br>CSeq: 3 ACK<br>User-Agent: XXXXX UserAgent/1.0<br>Content-Length: 0</p>
<p><br>--- (8 headers 0 lines)---<br>TestServer*CLI> <br><-- SIP read from <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>: <br>INVITE sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">
172.28.174.25:5060</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>
><br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>CSeq: 4 INVITE<br>Contact: <sip:XXXXXX@172.28.174.25:5060><br>Proxy-Authorization: Digest username="XXXXXX", realm="asterisk", nonce="452293c4", uri="
<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>", response="fd3b9b4d26c57a36ca4b1e96cca350ff", algorithm=MD5<br>Subject: no subject<br>User-Agent: XXXXX UserAgent/1.0<br>Max-Forwards: 70
<br>Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER<br>Content-Type: application/SDP<br>Accept: application/SDP, text/plain<br>Accept-Encoding: identity<br>Content-Length: 287</p>
<p>v=0<br>o=XXXXXX 11375833482 11375833482 IN IP4 <a href="http://172.28.174.25">172.28.174.25</a><br>s=VaxSoft Inc.<br>c=IN IP4 <a href="http://172.28.174.25">172.28.174.25</a><br>t=0 0<br>m=audio 15000 RTP/AVP 3 98 8 0 101
<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:98 iLBC/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16</p>
<p>--- (16 headers 12 lines)---<br>Using INVITE request as basis request - <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>Sending to <a href="http://172.28.174.25">172.28.174.25
</a> : 5060 (NAT)<br>Found user 'XXXXXX'<br>Found RTP audio format 3<br>Found RTP audio format 98<br>Found RTP audio format 8<br>Found RTP audio format 0<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://172.28.174.25:15000">
172.28.174.25:15000</a><br>Peer video RTP is at port <a href="http://172.28.174.25:65535">172.28.174.25:65535</a><br>Found description format GSM<br>Found description format iLBC<br>Found description format PCMA<br>Found description format PCMU
<br>Found description format telephone-event<br>Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc)<br>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
<br>Looking for 123456788 in prepaid (domain XXX.XXX.XX.XXX)<br>list_route: hop: <sip:XXXXXX@172.28.174.25:5060><br>Transmitting (NAT) to <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>:<br>SIP/2.0 100 Trying
<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060</a>;received=<a href="http://66.193.155.2">66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX
</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>><br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>
CSeq: 4 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX
</a>><br>Content-Length: 0</p>
<p><br>---<br>We're at XXX.XXX.XX.XXX port 17532<br>Video is at XXX.XXX.XX.XXX port 14338<br>Adding codec 0x400 (ilbc) to SDP<br>Adding codec 0x2 (gsm) to SDP<br>Adding codec 0x4 (ulaw) to SDP<br>Adding codec 0x8 (alaw) to SDP
<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (NAT) to <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060
</a>;received=<a href="http://66.193.155.2">66.193.155.2</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">
sip:123456788@XXX.XXX.XX.XXX</a>>;tag=as2894778b<br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>CSeq: 4 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
<br>Max-Forwards: 70<br>Contact: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>><br>Content-Type: application/sdp<br>Content-Length: 291<br>TestServer*CLI> <br>v=0<br>o=root 4095 4095 IN IP4
XXX.XXX.XX.XXX<br>s=session<br>c=IN IP4 XXX.XXX.XX.XXX<br>t=0 0<br>m=audio 17532 RTP/AVP 98 3 0 8 101<br>a=rtpmap:98 iLBC/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000
<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -</p>
<p>TestServer*CLI> <br><-- SIP read from <a href="http://66.193.155.2:46478">66.193.155.2:46478</a>: <br>ACK sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0<br>Via: SIP/2.0/UDP <a href="http://172.28.174.25:5060">172.28.174.25:5060
</a><br>From: XXXXXX <<a href="mailto:sip:XXXXXX@XXX.XXX.XX.XXX">sip:XXXXXX@XXX.XXX.XX.XXX</a>>;tag=174cf52<br>To: <<a href="mailto:sip:123456788@XXX.XXX.XX.XXX">sip:123456788@XXX.XXX.XX.XXX</a>>;tag=as2894778b
<br>Call-ID: <a href="mailto:7469ea-26568-43ce24f4@172.28.174.25">7469ea-26568-43ce24f4@172.28.174.25</a><br>CSeq: 4 ACK<br>User-Agent: XXXXX UserAgent/1.0<br>Content-Length: 0<br></p></div>
<div> </div>
<div> </div>
<div>Below is the RTP Logs:</div>
<div> </div>
<div>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37407, ts 878720, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5884, ts 897565, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37408, ts 878880, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5885, ts 897725, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37409, ts 879040, len 33)<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37410, ts 879200, len 33)
<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5886, ts 897885, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5887, ts 898045, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37411, ts 879360, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5888, ts 898205, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37412, ts 879520, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5889, ts 898365, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37413, ts 879680, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5890, ts 898525, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37414, ts 879840, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5891, ts 898685, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37415, ts 880000, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5892, ts 898845, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37416, ts 880160, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5893, ts 899005, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37417, ts 880320, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5894, ts 899165, len 33)
<br>Sent RTP packet to <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 37418, ts 880480, len 33)<br>Got RTP packet from <a href="http://66.193.155.2:15000">66.193.155.2:15000</a> (type 3, seq 5895, ts 899325, len 33)
</div>
<div> </div>
<div> </div>
<div> </div>
<div> </div>
<div>Regards,</div>
<div>Kengie</div>