<span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">hi,</span><div class="im" style="color:rgb(80,0,80);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
<div><br><span style="color:rgb(34,34,34)">"User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429)"</span></div><div><font color="#222222" face="arial, sans-serif"><br></font></div></div><div style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)">
<font color="#222222" face="arial, sans-serif">copy from asterisk 11 rtp.conf</font></div><div style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13px;background-color:rgb(255,255,255)"><font color="#222222" face="arial, sans-serif"><div>
rtpstart=10000</div><div>rtpend=20000</div><div><br></div><div>have you changed port range? if no then</div><div>your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port.</div>
<div>try to change rtpend=30000 or if there is option in softphone restrict it to use same range as in rtp.conf.</div><div><br></div><div>let me know if this solve you problem.</div></font></div><br><div class="gmail_quote">
On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad <span dir="ltr"><<a href="mailto:asghar144@gmail.com" target="_blank">asghar144@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
hi,<div class="im"><div><br><span style="color:rgb(34,34,34);font-size:13px;font-family:arial,sans-serif">"User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429)"</span></div><div>
<font color="#222222" face="arial, sans-serif"><br></font></div></div><div><font color="#222222" face="arial, sans-serif">copy from asterisk 11 rtp.conf</font></div><div><font color="#222222" face="arial, sans-serif"><div>
rtpstart=10000</div>
<div>rtpend=20000</div><div><br></div><div>have you changed port range? if no then</div><div>your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port.</div><div>try to change rtpend=30000 or if there is option in softphone restrict it to use same range as in rtp.conf.</div>
<div><br></div><div>let me know if this solve you problem.</div></font></div><div class="HOEnZb"><div class="h5"><div><font color="#222222" face="arial, sans-serif"><br></font></div><div><font color="#222222" face="arial, sans-serif"><br>
</font><div class="gmail_quote">
On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn <span dir="ltr"><<a href="mailto:mitch_ml@claborn.net" target="_blank">mitch_ml@claborn.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3.<br>
There is no NAT involved in the network at all (it is disabled in sip.conf).<br>
<br>
Here are the SIP messages capture via wireshark on the client during one problem call. Following these SIP messages, the wireshark trace shows only RTP packets from server (172.16.0.245) to client (172.16.0.71) except for an occasional RTCP packet from client to server (sample below).<br>
<br>
Any help is appreciated. The uses are really beating me up to get this fixed.<br>
<br>
--------------------<br>
<br>
INVITE <a href="http://sip:KWakmn@172.16.0.71:5060" target="_blank">sip:KWakmn@172.16.0.71:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=<u></u>z9hG4bK19e2246d<br>
Max-Forwards: 70<br>
From: <<a href="mailto:sip%3A2392230612@172.16.0.245" target="_blank">sip:2392230612@172.16.0.245</a>>;<u></u>tag=as4b489afc<br>
To: <<a href="http://sip:KWakmn@172.16.0.71:5060" target="_blank">sip:KWakmn@172.16.0.71:5060</a>><br>
Contact: <<a href="http://sip:2392230612@172.16.0.245:5060" target="_blank">sip:2392230612@172.16.0.245:<u></u>5060</a>><br>
Call-ID: <a href="http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060" target="_blank">52106f231b41169c7eabd3b43d0fc6<u></u>e8@172.16.0.245:5060</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 11.1.0<br>
Date: Tue, 19 Mar 2013 20:47:26 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>
X-mm-call: <a href="http://www.mcmurrayhatchery.com" target="_blank">http://www.mcmurrayhatchery.<u></u>com</a><br>
Content-Type: application/sdp<br>
Content-Length: 257<br>
<br>
v=0<br>
o=root 682517197 682517197 IN IP4 172.16.0.245<br>
s=Asterisk PBX 11.1.0<br>
c=IN IP4 172.16.0.245<br>
t=0 0<br>
m=audio 13428 RTP/AVP 0 8 101<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=ptime:20<br>
a=sendrecv<br>
<br>
------------------------------<u></u>-<br>
<br>
SIP/2.0 180 Ringing<br>
Via: SIP/2.0/UDP 172.16.0.245:5060;received=<u></u>172.16.0.245;branch=<u></u>z9hG4bK19e2246d<br>
Call-ID: <a href="http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060" target="_blank">52106f231b41169c7eabd3b43d0fc6<u></u>e8@172.16.0.245:5060</a><br>
From: <<a href="mailto:sip%3A2392230612@172.16.0.245" target="_blank">sip:2392230612@172.16.0.245</a>>;<u></u>tag=as4b489afc<br>
To: <<a href="mailto:sip%3AKWakmn@172.16.0.71" target="_blank">sip:KWakmn@172.16.0.71</a>>;tag=<u></u>7543f39a-7ca0-434b-8281-<u></u>e6dc2adc4aa3<br>
CSeq: 102 INVITE<br>
Contact: <<a href="http://sip:KWakmn@172.16.0.71:5060" target="_blank">sip:KWakmn@172.16.0.71:5060</a>><br>
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL<br>
Content-Length: 0<br>
<br>
------------------------------<u></u>-----------------------<br>
<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP 172.16.0.245:5060;received=<u></u>172.16.0.245;branch=<u></u>z9hG4bK19e2246d<br>
Call-ID: <a href="http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060" target="_blank">52106f231b41169c7eabd3b43d0fc6<u></u>e8@172.16.0.245:5060</a><br>
From: <<a href="mailto:sip%3A2392230612@172.16.0.245" target="_blank">sip:2392230612@172.16.0.245</a>>;<u></u>tag=as4b489afc<br>
To: <<a href="mailto:sip%3AKWakmn@172.16.0.71" target="_blank">sip:KWakmn@172.16.0.71</a>>;tag=<u></u>7543f39a-7ca0-434b-8281-<u></u>e6dc2adc4aa3<br>
CSeq: 102 INVITE<br>
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL<br>
Contact: <<a href="http://sip:KWakmn@172.16.0.71:5060" target="_blank">sip:KWakmn@172.16.0.71:5060</a>><br>
Supported: replaces, 100rel<br>
Content-Type: application/sdp<br>
Content-Length: 234<br>
<br>
v=0<br>
o=asset071 3572714846 1 IN IP4 172.16.0.71<br>
s=sflphone<br>
c=IN IP4 172.16.0.71<br>
t=0 0<br>
m=audio 39408 RTP/AVP 0<br>
a=rtpmap:0 PCMU/8000<br>
a=sendrecv<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=rtcp:39409 IN IP4 172.16.0.71<br>
<br>
------------------------------<u></u>-----------------<br>
<br>
ACK <a href="http://sip:KWakmn@172.16.0.71:5060" target="_blank">sip:KWakmn@172.16.0.71:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 172.16.0.245:5060;branch=<u></u>z9hG4bK289d6da2<br>
Max-Forwards: 70<br>
From: <<a href="mailto:sip%3A2392230612@172.16.0.245" target="_blank">sip:2392230612@172.16.0.245</a>>;<u></u>tag=as4b489afc<br>
To: <<a href="http://sip:KWakmn@172.16.0.71:5060" target="_blank">sip:KWakmn@172.16.0.71:5060</a>>;<u></u>tag=7543f39a-7ca0-434b-8281-<u></u>e6dc2adc4aa3<br>
Contact: <<a href="http://sip:2392230612@172.16.0.245:5060" target="_blank">sip:2392230612@172.16.0.245:<u></u>5060</a>><br>
Call-ID: <a href="http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060" target="_blank">52106f231b41169c7eabd3b43d0fc6<u></u>e8@172.16.0.245:5060</a><br>
CSeq: 102 ACK<br>
User-Agent: Asterisk PBX 11.1.0<br>
Content-Length: 0<br>
<br>
------------------------------<u></u>------------------------------<br>
<br>
SAMPLE RTCP packet from client to server<br>
<br>
No. Time Source Destination Protocol Length Info<br>
240 15:47:39.965483 172.16.0.71 172.16.0.245 RTCP 102 Receiver Report Source description<br>
<br>
Frame 240: 102 bytes on wire (816 bits), 102 bytes captured (816 bits)<br>
Ethernet II, Src: Dell_e7:fc:b0 (00:25:64:e7:fc:b0), Dst: 90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35)<br>
Internet Protocol Version 4, Src: 172.16.0.71 (172.16.0.71), Dst: 172.16.0.245 (172.16.0.245)<br>
User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429)<br>
Real-time Transport Control Protocol (Receiver Report)<br>
[Stream setup by SDP (frame 36)]<br>
[Setup frame: 36]<br>
[Setup Method: SDP]<br>
10.. .... = Version: RFC 1889 Version (2)<br>
..0. .... = Padding: False<br>
...0 0001 = Reception report count: 1<br>
Packet type: Receiver Report (201)<br>
Length: 7 (32 bytes)<br>
Sender SSRC: 0x841ef2ea (2216620778)<br>
Source 1<br>
Identifier: 0x28bcc3a6 (683459494)<br>
SSRC contents<br>
Fraction lost: 254 / 256<br>
Cumulative number of packets lost: 37134<br>
Extended highest sequence number received: 37331<br>
Sequence number cycles count: 0<br>
Highest sequence number received: 37331<br>
Interarrival jitter: 160008128<br>
Last SR timestamp: 0 (0x00000000)<br>
Delay since last SR timestamp: 0 (0 milliseconds)<br>
Real-time Transport Control Protocol (Source description)<br>
[Stream setup by SDP (frame 36)]<br>
[Setup frame: 36]<br>
[Setup Method: SDP]<br>
10.. .... = Version: RFC 1889 Version (2)<br>
..0. .... = Padding: False<br>
...0 0001 = Source count: 1<br>
Packet type: Source description (202)<br>
Length: 6 (28 bytes)<br>
Chunk 1, SSRC/CSRC 0x841EF2EA<br>
Identifier: 0x841ef2ea (2216620778)<br>
SDES items<br>
Type: CNAME (user and domain) (1)<br>
Length: 17<br>
Text: kristin@localhost<br>
Type: END (0)<br>
[RTCP frame length check: OK - 60 bytes]<br>
<br>
<br>
<br>
<br>
<br>
Mitch<br>
<br></blockquote></div></div></div></div></blockquote></div>