silence on both ends

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yannick
Posts: 6
Joined: Thu Sep 01, 2011 2:14 pm

silence on both ends

Post by yannick » Fri Sep 28, 2012 8:58 am

I'm using this ruby code to first call my softphone and if it's not taken within 10 seconds to redirect to my mobile phone and finaly to my voicemail if none is taken:

Code: Select all

if sys.In then
       sys.ExtendScriptTimeout(40) # Extends the time script will survive without getting a trying response from a SIP server from 10 to 40s. 

       sys.Dial("yannickmobiel@intervoip.com&0031653667420@Intervoip[dt=10,fu=#{@cid}]&17772098227@in.callcentric.com[dt=35]") 

end
The problem is that if I pick up the phone on my softphone after about 10 seconds (not if i pickup immediately) both ends hear nothing.
Is there something I can do about this problem?

Also it seems if the dialplan hasn't been used for a while (like 2 hours) when somebody calls it takes forever (like 10s) before a ringtone is heard. Not sure where to look for the solution.

User avatar
Aaron
Site Admin
Posts: 4563
Joined: Thu Jul 12, 2007 12:13 am

Re: silence on both ends

Post by Aaron » Fri Sep 28, 2012 9:43 am

When you have the audio issue is the call going to a SIP device or to a mobile phone?

As far as the delay in your ringing time turn on dialplan tracing by putting your email address in the "Trace Email Address" box. You'll then get an email each time your dialplan executes. When you have a call with the ringing delay post the contents of the email you get up here and we can diagnose it.

yannick
Posts: 6
Joined: Thu Sep 01, 2011 2:14 pm

Re: silence on both ends

Post by yannick » Fri Sep 28, 2012 10:39 am

Happens both on SIP and mobile device.

Will post trace log here when I get the behavior again (can't test now as it seems not to delay now).

Do you want me to post the log for a no audio call?

User avatar
Aaron
Site Admin
Posts: 4563
Joined: Thu Jul 12, 2007 12:13 am

Re: silence on both ends

Post by Aaron » Sat Sep 29, 2012 3:18 am

Yes it would be worth posting a trace for you no audio call. I suspect it won't reveal the problem since audio issues are mostly outside of the SIP layer but it won't hurt to check.

yannick
Posts: 6
Joined: Thu Sep 01, 2011 2:14 pm

Re: silence on both ends

Post by yannick » Mon Oct 01, 2012 3:41 pm

here is the log for the no audio call.

Code: Select all

SIPTransaction=> SIPTransaction=>Request received udp:67.222.131.147:5070<-udp:67.222.131.147:5060
INVITE sip:yannick@sipsorcery.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5060;branch=z9hG4bK4e37fc7cde34817cccdc10ccb9ff3c2654a1e470;rport=5060;received=67.222.131.147
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK90f5.10dd6ce5.0;received=83.143.188.165;rport=5060
Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK1834862955
To: <sip:31707113181@sip1.budgetphone.nl;user=phone>
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1723758906
Call-ID: 1393956502@83.143.188.161
CSeq: 20 INVITE
Contact: <sip:SIP_5Fc@83.143.188.161:5060;bnat=yes>
Max-Forwards: 11
Record-Route: <sip:83.143.188.165;lr;ftag=1723758906>
Content-Length: 319
Content-Type: application/sdp
Proxy-ReceivedFrom: udp:83.143.188.165:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER, PRACK, OPTIONS, REFER, SUBSCRIBE, NOTIFY
P-Asserted-Identity: <sip:0703226966@83.143.188.161;user=phone>

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

DialPlan=> Dialplan trace commenced at 28 Sep 2012 03:38:20:032.
DialPlan=> Commencing Dial with: yannickmobiel@intervoip.com&0031653667420@Intervoip[dt=10,fu=]&17772098227@in.callcentric.com[dt=35].
DialPlan=> ForkCall commencing call leg to sip:yannickmobiel@intervoip.com.
DialPlan=> ForkCall commencing call leg to sip:0031653667420@sip.intervoip.com.
DialPlan=> Switching to sip:yannickmobiel@intervoip.com:5060 via udp:67.222.131.147:5060.
DialPlan=> ForkCall commencing call leg to sip:17772098227@in.callcentric.com.
DialPlan=> Delaying call leg to sip:0031653667420@sip.intervoip.com by 10s.
DialPlan=> SDP on UAC call had public IP not mangled, RTP socket 83.143.188.165:41402.
DialPlan=> Delaying call leg to sip:17772098227@in.callcentric.com by 35s.
SIPTransaction=> Send Request reliable udp:67.222.131.147:5070->udp:67.222.131.147:5060
INVITE sip:yannickmobiel@intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bKa8f0c120e9e04ca28032a7835dfe4b86;rport
To: <sip:yannickmobiel@intervoip.com>
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1298563943
Call-ID: 2ffb1bb8c7c7481983002dee2666010d
CSeq: 1 INVITE
Contact: <sip:Anonymous@67.222.131.147:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Length: 319
Content-Type: application/sdp

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bKa8f0c120e9e04ca28032a7835dfe4b86;rport
To: <sip:yannickmobiel@intervoip.com>
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1298563943
Call-ID: 2ffb1bb8c7c7481983002dee2666010d
CSeq: 1 INVITE
Contact: <sip:77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


DialPlan=> Information response 100 Trying for sip:yannickmobiel@intervoip.com.
SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bKa8f0c120e9e04ca28032a7835dfe4b86;rport
To: <sip:yannickmobiel@intervoip.com>
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1298563943
Call-ID: 2ffb1bb8c7c7481983002dee2666010d
CSeq: 1 INVITE
Contact: <sip:77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


DialPlan=> Information response 180 Ringing for sip:yannickmobiel@intervoip.com.
SIPTransaction=> Send Info Response udp:67.222.131.147:5070->udp:67.222.131.147:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.222.131.147:5060;branch=z9hG4bK4e37fc7cde34817cccdc10ccb9ff3c2654a1e470;rport=5060;received=67.222.131.147
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK90f5.10dd6ce5.0;received=83.143.188.165;rport=5060
Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK1834862955
To: <sip:31707113181@sip1.budgetphone.nl;user=phone>;tag=1355045985
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1723758906
Call-ID: 1393956502@83.143.188.161
CSeq: 20 INVITE
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE
Content-Length: 0


SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bKa8f0c120e9e04ca28032a7835dfe4b86;rport
To: <sip:yannickmobiel@intervoip.com>
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1298563943
Call-ID: 2ffb1bb8c7c7481983002dee2666010d
CSeq: 1 INVITE
Contact: <sip:77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


DialPlan=> Information response 180 Ringing for sip:yannickmobiel@intervoip.com.
DialPlan=> SIPClientUserAgent Call using alternate outbound proxy of udp:67.222.131.147:5060.
DialPlan=> Switching to sip:0031653667420@sip.intervoip.com:5060 via udp:67.222.131.147:5060.
DialPlan=> SDP on UAC call had public IP not mangled, RTP socket 83.143.188.165:41402.
SIPTransaction=> Send Request reliable udp:67.222.131.147:5070->udp:67.222.131.147:5060
INVITE sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK10d15b10bf524160bcf258a08b2ca108;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 1 INVITE
Contact: <sip:+31707113181@67.222.131.147:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Length: 319
Content-Type: application/sdp
Proxy-SendFrom: udp:67.222.131.147:5060

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

SIPTransaction=> Send Request retransmit 2 udp:67.222.131.147:5070->udp:67.222.131.147:5060
INVITE sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK10d15b10bf524160bcf258a08b2ca108;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 1 INVITE
Contact: <sip:+31707113181@67.222.131.147:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Content-Length: 319
Content-Type: application/sdp
Proxy-SendFrom: udp:67.222.131.147:5060

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK10d15b10bf524160bcf258a08b2ca108;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 1 INVITE
Contact: <sip:0031653667420@77.72.169.134:5060>
WWW-Authenticate: Digest realm="sip.intervoip.com",nonce="1836538344",algorithm=MD5
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


SIPTransaction=> Send Request udp:67.222.131.147:5070->udp:67.222.131.147:5060
ACK sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK10d15b10bf524160bcf258a08b2ca108;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
Proxy-SendFrom: udp:67.222.131.147:5060


DialPlan=> Response 401 Unauthorized for sip:0031653667420@sip.intervoip.com.
SIPTransaction=> Send Request reliable udp:67.222.131.147:5070->udp:67.222.131.147:5060
INVITE sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 INVITE
Contact: <sip:+31707113181@67.222.131.147:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Authorization: Digest username="yannickmobiel",realm="sip.intervoip.com",nonce="1836538344",uri="sip:0031653667420@sip.intervoip.com",response="74defaec4fbdd87945d8da7001506338",algorithm=MD5
Content-Length: 319
Content-Type: application/sdp
Proxy-SendFrom: udp:67.222.131.147:5060

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

SIPTransaction=> Send Request retransmit 2 udp:67.222.131.147:5070->udp:67.222.131.147:5060
INVITE sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 INVITE
Contact: <sip:+31707113181@67.222.131.147:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Authorization: Digest username="yannickmobiel",realm="sip.intervoip.com",nonce="1836538344",uri="sip:0031653667420@sip.intervoip.com",response="74defaec4fbdd87945d8da7001506338",algorithm=MD5
Content-Length: 319
Content-Type: application/sdp
Proxy-SendFrom: udp:67.222.131.147:5060

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

SIPTransaction=> Send Request retransmit 3 udp:67.222.131.147:5070->udp:67.222.131.147:5060
INVITE sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 INVITE
Contact: <sip:+31707113181@67.222.131.147:5070>
Max-Forwards: 70
User-Agent: www.sipsorcery.com
Authorization: Digest username="yannickmobiel",realm="sip.intervoip.com",nonce="1836538344",uri="sip:0031653667420@sip.intervoip.com",response="74defaec4fbdd87945d8da7001506338",algorithm=MD5
Content-Length: 319
Content-Type: application/sdp
Proxy-SendFrom: udp:67.222.131.147:5060

v=0
o=SIP_5Fc 123456 654321 IN IP4 83.143.188.165
s=-
c=IN IP4 83.143.188.165
t=0 0
m=audio 41402 RTP/AVP 8 0 4 18 3 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:3 GSM/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15
a=nortpproxy:yes

SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bKa8f0c120e9e04ca28032a7835dfe4b86;rport
To: <sip:yannickmobiel@intervoip.com>;tag=780313ac500fdcd810dd324
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1298563943
Call-ID: 2ffb1bb8c7c7481983002dee2666010d
CSeq: 1 INVITE
Contact: <sip:77.72.169.134:5060>
Content-Length: 202
Content-Type: application/sdp
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

v=0
o=CARRIER 1348828712 1348828712 IN IP4 77.72.168.99
s=SIP Call
c=IN IP4 77.72.168.99
t=0 0
m=audio 11416 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=ptime:20

SIPTransaction=> Send Request udp:67.222.131.147:5070->udp:67.222.131.147:5060
ACK sip:77.72.169.134:5060 SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK11384ccb1a1242ab8b35424fe4a05270;rport
To: <sip:yannickmobiel@intervoip.com>;tag=780313ac500fdcd810dd324
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1298563943
Call-ID: 2ffb1bb8c7c7481983002dee2666010d
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0


DialPlan=> Response 200 Ok for sip:yannickmobiel@intervoip.com.
DialPlan=> SDP on UAC response had public IP not mangled, RTP socket 77.72.168.99:11416.
DialPlan=> Cancelling all call legs for ForkCall app.
DialPlan=> Cancelling forwarded call leg, sending CANCEL to sip:0031653667420@sip.intervoip.com.
SIPTransaction=> Send Final Response Reliable udp:67.222.131.147:5070->67.222.131.147:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.222.131.147:5060;branch=z9hG4bK4e37fc7cde34817cccdc10ccb9ff3c2654a1e470;rport=5060;received=67.222.131.147
Via: SIP/2.0/UDP 83.143.188.165;branch=z9hG4bK90f5.10dd6ce5.0;received=83.143.188.165;rport=5060
Via: SIP/2.0/UDP 83.143.188.161:5060;rport=5060;received=83.143.188.161;branch=z9hG4bK1834862955
To: <sip:31707113181@sip1.budgetphone.nl;user=phone>;tag=1355045985
From: <sip:0703226966@83.143.188.161;user=phone>;tag=1723758906
Call-ID: 1393956502@83.143.188.161
CSeq: 20 INVITE
Contact: <sip:67.222.131.147:5070>
Record-Route: <sip:83.143.188.165;lr;ftag=1723758906>
Server: www.sipsorcery.com
Content-Length: 202
Content-Type: application/sdp

v=0
o=CARRIER 1348828712 1348828712 IN IP4 77.72.168.99
s=SIP Call
c=IN IP4 77.72.168.99
t=0 0
m=audio 11416 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=ptime:20

SIPTransaction=> Send Request reliable udp:67.222.131.147:5070->udp:67.222.131.147:5060
CANCEL sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 CANCEL
Max-Forwards: 70
Content-Length: 0
Proxy-SendFrom: udp:67.222.131.147:5060


DialPlan=> Dial command was successfully answered in 12.72s.
DialPlan=> Dialplan cleanup for yannick.
SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 INVITE
Contact: <sip:0031653667420@77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


DialPlan=> Information response 100 Trying for sip:0031653667420@sip.intervoip.com.
DialPlan=> Call sip:0031653667420@sip.intervoip.com has already been cancelled once, trying again.
SIPTransaction=> Send Request udp:67.222.131.147:5070->udp:67.222.131.147:5060
CANCEL sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 CANCEL
Max-Forwards: 70
Content-Length: 0
Proxy-SendFrom: udp:67.222.131.147:5060


SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>;tag=780313ac500fdcd810dd326
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 INVITE
Contact: <sip:0031653667420@77.72.169.134:5060>
Content-Length: 208
Content-Type: application/sdp
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE

v=0
o=yannickmobiel 1348828712 1348828712 IN IP4 77.72.168.73
s=SIP Call
c=IN IP4 77.72.168.73
t=0 0
m=audio 14116 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=ptime:20

DialPlan=> Information response 183 Session progress for sip:0031653667420@sip.intervoip.com.
DialPlan=> Call sip:0031653667420@sip.intervoip.com has already been cancelled once, trying again.
SIPTransaction=> Send Request udp:67.222.131.147:5070->udp:67.222.131.147:5060
CANCEL sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 CANCEL
Max-Forwards: 70
Content-Length: 0
Proxy-SendFrom: udp:67.222.131.147:5060


SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 CANCEL
Contact: <sip:0031653667420@77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


SIPTransaction=> Received Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 487 Request terminated
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 INVITE
Contact: <sip:0031653667420@77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


SIPTransaction=> Send Request udp:67.222.131.147:5070->udp:67.222.131.147:5060
ACK sip:0031653667420@sip.intervoip.com SIP/2.0
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 ACK
Max-Forwards: 70
Authorization: Digest username="yannickmobiel",realm="sip.intervoip.com",nonce="1836538344",uri="sip:0031653667420@sip.intervoip.com",response="74defaec4fbdd87945d8da7001506338",algorithm=MD5
Content-Length: 0
Proxy-SendFrom: udp:67.222.131.147:5060


DialPlan=> Response 487 Request terminated for sip:0031653667420@sip.intervoip.com.
SIPTransaction=> Received Duplicate Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 CANCEL
Contact: <sip:0031653667420@77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


SIPTransaction=> Received Duplicate Response udp:67.222.131.147:5070<-udp:67.222.131.147:5060
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 67.222.131.147:5070;branch=z9hG4bK6a3a846d039248fb8203a6ec576bd71e;rport
To: <sip:0031653667420@sip.intervoip.com>
From: <sip:+31707113181@sip.intervoip.com>;tag=1107423766
Call-ID: b54267aa1cd445079fdf935ea2e1c13b
CSeq: 2 CANCEL
Contact: <sip:0031653667420@77.72.169.134:5060>
Content-Length: 0
Proxy-ReceivedFrom: udp:77.72.169.134:5060
Proxy-ReceivedOn: udp:67.222.131.147:5060
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE


DialPlan=> Dialplan trace completed at 28 Sep 2012 03:39:04:782.

User avatar
Aaron
Site Admin
Posts: 4563
Joined: Thu Jul 12, 2007 12:13 am

Re: silence on both ends

Post by Aaron » Wed Oct 03, 2012 9:27 am

One thing I spotted is that on the INVITE request from budgetphone this line is in the call payload "a=nortpproxy:yes". That would seemingly indicate that budgetphone are not proxying the audio on the call. Given that you've got an audio issue it would be good if you were able to try when budgetphone do proxy the audio. As such it would be worth logging into your budgetphone account and seeing if there is any setting relating to audio or proxying or sometimes it is called something like "symmetric NAT handling".

yannick
Posts: 6
Joined: Thu Sep 01, 2011 2:14 pm

Re: silence on both ends

Post by yannick » Mon Oct 08, 2012 9:23 am

got in contact with budgetphone.nl and this was their reply:

Because Sipsorcery can be reached by internet it is not necessary.

to technical for me, but anyway there is no setting to adjust it at their end.

User avatar
Aaron
Site Admin
Posts: 4563
Joined: Thu Jul 12, 2007 12:13 am

Re: silence on both ends

Post by Aaron » Tue Oct 09, 2012 10:56 am

That response does make sense. Because the sipsorcery server uses a public static IP address there is no NAT involved. However that premise breaks down when the destination the sipsorcery forwards to is behind a NAT.

I've had another look at your original post and it occurred to me that maybe if you take too long to answer the call on your softphone one of your subsequent call legs answers the call and messes around with the audio. Can you do a test where you just have a single leg in your dial string which forwards to your Bria softphone and see if you get the same issue.

yannick
Posts: 6
Joined: Thu Sep 01, 2011 2:14 pm

Re: silence on both ends

Post by yannick » Tue Oct 09, 2012 2:18 pm

When I remove the other 2 legs the problem ceases.
Problem only arises when I pick up the phone just around the time that the second leg (a conventional mobile line) is called.

User avatar
Aaron
Site Admin
Posts: 4563
Joined: Thu Jul 12, 2007 12:13 am

Re: silence on both ends

Post by Aaron » Thu Oct 11, 2012 9:49 am

Ok I think I've at least a better grasp on the problem now. Sorry I didn't get it at the start and assumed it was a NAT problem somewhere as that's the cause of 99% of audio issues with SIP calls.

Unfortunately it still doesn't help me identify what the problem could be. One thing I think would be worth trying is to see if using a different provider for your second call leg to your mobile phone makes any difference (even if it's just a different Betamax provider). For example:

Code: Select all

sys.Dial("yannickmobiel@intervoip.com&0031653667420@different_provider[dt=10,fu=#@cid}]&17772098227@in.callcentric.com[dt=35]") 
There's a possibility the intervoip server could be confused with two simultaneous calls on the same account from the same IP address and messes up the audio channels.

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