<div>Hi,</div><div><br></div><div>Being audible sometime or bad voice quality is only due to internet latency or bad internet situation.</div><div><br></div><font size="1"><span style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)">[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: Retransmission timeout reached on transmission </span><a href="http://551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060/" target="_blank" style="color:rgb(17,85,204);font-family:arial,sans-serif;background-color:rgb(255,255,255)">551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060</a><span style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)"> for seqno 102 (Critical Request) -- See </span><a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank" style="color:rgb(17,85,204);font-family:arial,sans-serif;background-color:rgb(255,255,255)">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)">
<span style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)">Packet timed out after 32000ms with no response</span><br style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)">
<span style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)">[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up call </span><a href="http://551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060/" target="_blank" style="color:rgb(17,85,204);font-family:arial,sans-serif;background-color:rgb(255,255,255)">551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060</a><span style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)"> - no reply to our critical packet (see </span><a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank" style="color:rgb(17,85,204);font-family:arial,sans-serif;background-color:rgb(255,255,255)">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><span style="color:rgb(34,34,34);font-family:arial,sans-serif;background-color:rgb(255,255,255)">).</span> </font><br>
<br>The above lines again telling that there is some problem sending sequential packets to some endpoint. That may lead to disconnection of call after some time..as it is currently doing so.<div><br></div><div>Try setting some more NAT parameters...as you said localnet. Set <b>localnet=</b> parameter entries in your asterisk server sip configurations.</div>
<div><br></div><div>BR</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Wed, Jul 4, 2012 at 2:13 PM, alok srivastava <span dir="ltr"><<a href="mailto:alokkic@gmail.com" target="_blank">alokkic@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<p style="margin-bottom:0cm;font-weight:normal" align="LEFT"><font size="3">thanks
Samy<br>i have set nat=yes, now getting sound from both side but
there is too uch disturbance. soetime we becoe audible and
sometime not.i did not set extern ip coz my asterisk server is
directly configured on public ip. I have softphones on some where
localnets separate from asterisk server campus . i also set "sip set debug on" CLI prompt. this is giving following error.</font></p><p style="margin-bottom:0cm;font-weight:normal" align="LEFT"><font size="3">when i test sip traffic on wireshark "401 unauthorize" error getting this error cli prompt also showing.</font></p>
<p style="margin-bottom:0cm;font-weight:normal" align="LEFT"><font size="3">my first softph(9001) is on localnet 192.168.1.136 and 2nd softphone (9000) in another localnet in another campus(192.168.6.25) <br></font></p>
<br><br>Scheduling destruction of SIP dialog '<a href="http://551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060" target="_blank">551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060</a>' in 32000 ms (Method: INVITE)<br>
[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3660 retrans_pkt: Retransmission timeout reached on transmission <a href="http://551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060" target="_blank">551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060</a> for seqno 102 (Critical Request) -- See <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
Packet timed out after 32000ms with no response<br>[Jul 4 19:37:39] WARNING[3054]: chan_sip.c:3689 retrans_pkt: Hanging up call <a href="http://551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060" target="_blank">551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060</a> - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions" target="_blank">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<br>
-- SIP/9000-00000005 is circuit-busy<br> == Everyone is busy/congested at this time (1:0/1/0)<br> -- Auto fallthrough, channel 'SIP/9001-00000004' status is 'CONGESTION'<br><br><--- Reliably Transmitting (NAT) to <a href="http://122.163.193.94:1801" target="_blank">122.163.193.94:1801</a> ---><br>
SIP/2.0 503 Service Unavailable<br>Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;received=122.163.193.94;rport=1801<br>From: "9001"<<a href="mailto:sip%3A9001@122.160.154.189" target="_blank">sip:9001@122.160.154.189</a>>;tag=b0785362<br>
To: <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=as6c7d28d1<br>Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.<br>CSeq: 2 INVITE<br>Server: Asterisk PBX 10.0.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>
Supported: replaces, timer<br>X-Asterisk-HangupCause: No user responding<br>X-Asterisk-HangupCauseCode: 18<br>Content-Length: 0<br><br><br><------------><br>Really destroying SIP dialog '<a href="http://551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060" target="_blank">551b9e744a6b41ee2c00033e22b333c0@122.160.154.189:5060</a>' Method: INVITE<br>
<br><--- SIP read from UDP:<a href="http://122.163.193.94:1801" target="_blank">122.163.193.94:1801</a> ---><br>ACK <a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK-d8754z-db50ccbe7eb74a0a-1---d8754z-;rport<br>
Max-Forwards: 70<br>To: <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=as6c7d28d1<br>From: "9001"<<a href="mailto:sip%3A9001@122.160.154.189" target="_blank">sip:9001@122.160.154.189</a>>;tag=b0785362<br>
Call-ID: MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.<br>CSeq: 2 ACK<br>Content-Length: 0<br><br><-------------><br>--- (8 headers 0 lines) ---<br>Really destroying SIP dialog 'MGViNzg2YjNiOGQ4YzRmNDgxYmU2OTc2NDk4YjA1NGU.' Method: ACK<br>
<br><--- SIP read from UDP:<a href="http://122.163.193.94:1801" target="_blank">122.163.193.94:1801</a> ---><br><br><br><-------------><br><br><--- SIP read from UDP:<a href="http://115.249.67.250:5060" target="_blank">115.249.67.250:5060</a> ---><br>
REGISTER sip:122.160.154.189 SIP/2.0<br>Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKskzxkdlp<br>Max-Forwards: 70<br>To: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>><br>
From: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=jcysf<br>Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110<br>CSeq: 954 REGISTER<br>Contact: <<a href="mailto:sip%3A9000@192.168.6.25" target="_blank">sip:9000@192.168.6.25</a>>;expires=3600<br>
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE<br>User-Agent: Twinkle/1.4.2<br>Content-Length: 0<br><br><-------------><br>--- (11 headers 0 lines) ---<br>Sending to <a href="http://115.249.67.250:5060" target="_blank">115.249.67.250:5060</a> (NAT)<br>
<br><--- Transmitting (NAT) to <a href="http://115.249.67.250:5060" target="_blank">115.249.67.250:5060</a> ---><br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKskzxkdlp;received=115.249.67.250;rport=5060<br>
From: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=jcysf<br>To: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=as26d4cd86<br>
Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110<br>CSeq: 954 REGISTER<br>Server: Asterisk PBX 10.0.0<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="278a3764"<br>
Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER)<br><br><--- SIP read from UDP:<a href="http://115.249.67.250:5060" target="_blank">115.249.67.250:5060</a> ---><br>
REGISTER sip:122.160.154.189 SIP/2.0<br>Via: SIP/2.0/UDP 192.168.6.25;rport;branch=z9hG4bKjmqlllxn<br>Max-Forwards: 70<br>To: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>><br>
From: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=jcysf<br>Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110<br>CSeq: 955 REGISTER<br>Contact: <<a href="mailto:sip%3A9000@192.168.6.25" target="_blank">sip:9000@192.168.6.25</a>>;expires=3600<br>
Authorization: Digest username="9000",realm="asterisk",nonce="278a3764",uri="sip:122.160.154.189",response="c7a119185514202d5f9cc10a86a93607",algorithm=MD5<br>Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE<br>
User-Agent: Twinkle/1.4.2<br>Content-Length: 0<br><br><-------------><br>--- (12 headers 0 lines) ---<br>Sending to <a href="http://115.249.67.250:5060" target="_blank">115.249.67.250:5060</a> (NAT)<br><br><--- Transmitting (NAT) to <a href="http://115.249.67.250:5060" target="_blank">115.249.67.250:5060</a> ---><br>
SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 192.168.6.25;branch=z9hG4bKjmqlllxn;received=115.249.67.250;rport=5060<br>From: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=jcysf<br>
To: "shekhar" <<a href="mailto:sip%3A9000@122.160.154.189" target="_blank">sip:9000@122.160.154.189</a>>;tag=as26d4cd86<br>Call-ID: qajoimvsihxpsff@alok-Inspiron-N5110<br>CSeq: 955 REGISTER<br>Server: Asterisk PBX 10.0.0<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH<br>Supported: replaces, timer<br>Expires: 3600<br>Contact: <<a href="mailto:sip%3A9000@192.168.6.25" target="_blank">sip:9000@192.168.6.25</a>>;expires=3600<br>
Date: Wed, 04 Jul 2012 14:08:17 GMT<br>Content-Length: 0<br><br><br><------------><br>Scheduling destruction of SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' in 32000 ms (Method: REGISTER)<br><br><--- SIP read from UDP:<a href="http://122.163.193.94:1801" target="_blank">122.163.193.94:1801</a> ---><br>
<br><br><-------------><br>Really destroying SIP dialog 'qajoimvsihxpsff@alok-Inspiron-N5110' Method: REGISTER<br><br><br>regards<br>abhi<div class="HOEnZb"><div class="h5"><br><br><br><br><br><br><br><br><br>
<br><div class="gmail_quote">On Mon, Jul 2, 2012 at 5:22 PM, SamyGo <span dir="ltr"><<a href="mailto:govoiper@gmail.com" target="_blank">govoiper@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
actually its a one-way audio issue due to NAT ! <div><br><div>alok , please explain your network flow for end to end client-server-client.</div><div><br>You may need to set nat=yes for your sip peer behind NAT. If the server is behind NAT router/firewall use externip=<public.ip.of.server> field.</div>
<div>Also provide sip traces of this call.</div><div>Another thing to do for your learning. Execute wireshark on both softphone systems and set "sip | rtp" as filter and see where are the RTP streams going on each end !</div>
<div><br></div><div>Take a complete capture on Asterisk server by executing the command "sip set debug on" and make a call.</div><div><br></div><div>BR</div><div>Sammy</div><div><div><div><br></div><div>
<br></div><div><div class="gmail_quote">
On Mon, Jul 2, 2012 at 4:39 PM, Thomas Kenyon <span dir="ltr"><<a href="mailto:digium@sanguinarius.co.uk" target="_blank">digium@sanguinarius.co.uk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div>alok srivastava wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
dear<br>
i have configured properly asterisk. At the one end i am using x-lite soft ph and another end twinkle. call is going properly from both end but after picking the phone not able to listen other one.<br>
when i checked the port 5060 on the asterisk server it is always showing closed while i have flushed all the rules from iptables (iptables -F)<br>
<br>
PORT STATE SERVICE VERSION<br>
5060/tcp closed sip<br>
<br>
telnet localhost 5060 (could not connect)<br>
<br>
regards<br>
alok<br>
<br>
<br>
</blockquote></div></div>
SIP is only used to setup (and stop etc.) the call. The actual audio is sent via rtp.<br>
<br>
/etc/asterisk/rtp.conf<br>
<br>
Should tell which ports asterisk is using for rtp, you will need to make sure that the remote host can connect to these ports.<br>
<br>
There are lots of articles around on how to resolve this.<div><div><br>
<br>
<br>
<br>
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