Dear All,
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.
Asterisk Version is 1.2.1
Asterisk RTP Range is 10000 to 20000
UA Listen RTP Port is 15000
Below is the the SIP Logs
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
CSeq: 1 REGISTER
Contact: "XXXXXX" <sip:XXXXXX@172.28.174.25:5060>
User-Agent: XXXXX UserAgent/1.0
Expires: 5000
Max-Forwards: 70
Content-Length: 0
--- (11 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 172.28.174.25 : 5060 (non-NAT)
Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:XXXXXX@XXX.XXX.XX.XXX>
Content-Length: 0
---
Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=as504de7b8
Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:XXXXXX@XXX.XXX.XX.XXX>
WWW-Authenticate: Digest realm="asterisk", nonce="31e7aa76"
Content-Length: 0
---
Scheduling destruction of call '74494a-1654e-43ce24ec@XXX.XXX.XX.XXX' in
15000 ms
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
REGISTER sip:XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
CSeq: 2 REGISTER
Contact: "XXXXXX" <sip:XXXXXX@172.28.174.25:5060>
Authorization: Digest username="XXXXXX", realm="asterisk",
nonce="31e7aa76",
uri="sip:XXX.XXX.XX.XXX",
response="3e54ea5e3a3b6df1e5db7b4a3182e18f",
algorithm=MD5
User-Agent: XXXXX UserAgent/1.0
Expires: 5000
Max-Forwards: 70
Content-Length: 0
--- (12 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 172.28.174.25 : 5060 (NAT)
Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:XXXXXX@XXX.XXX.XX.XXX>
Content-Length: 0
---
Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=as504de7b8
Call-ID: 74494a-1654e-43ce24ec@XXX.XXX.XX.XXX
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 3600
Contact: <sip:XXXXXX@172.28.174.25:5060>;expires=3600
Date: Wed, 18 Jan 2006 11:20:15 GMT
Content-Length: 0
---
Scheduling destruction of call '74494a-1654e-43ce24ec@XXX.XXX.XX.XXX' in
15000 ms
Destroying call '4CEFE29BD3864FFFA788C2D43F4E6FBD@wawan'
TestServer*CLI>
<-- SIP read from 202.83.167.103:30928:
--- (0 headers 0 lines) Nat keepalive ---
TestServer*CLI>
<-- SIP read from 218.111.26.5:5060:
--- (0 headers 0 lines) Nat keepalive ---
Destroying call 'IvDHn41FxnNwCePq@192.168.1.35'
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
INVITE sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 3 INVITE
Contact: <sip:XXXXXX@172.28.174.25:5060>
Subject: no subject
User-Agent: XXXXX UserAgent/1.0
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Type: application/SDP
Accept: application/SDP, text/plain
Accept-Encoding: identity
Content-Length: 287
v=0
o=XXXXXX 11375833482 11375833482 IN IP4 172.28.174.25
s=VaxSoft Inc.
c=IN IP4 172.28.174.25
t=0 0
m=audio 15000 RTP/AVP 3 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--- (15 headers 12 lines)---
Using INVITE request as basis request - 7469ea-26568-43ce24f4@172.28.174.25
Sending to 172.28.174.25 : 5060 (non-NAT)
Reliably Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>;tag=as733e9adc
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:123456788@XXX.XXX.XX.XXX>
Proxy-Authenticate: Digest realm="asterisk",
nonce="452293c4"
Content-Length: 0
---
Scheduling destruction of call '7469ea-26568-43ce24f4@172.28.174.25' in
15000 ms
Found user 'XXXXXX'
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
ACK sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>;tag=as733e9adc
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 3 ACK
User-Agent: XXXXX UserAgent/1.0
Content-Length: 0
--- (8 headers 0 lines)---
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
INVITE sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 4 INVITE
Contact: <sip:XXXXXX@172.28.174.25:5060>
Proxy-Authorization: Digest username="XXXXXX",
realm="asterisk",
nonce="452293c4", uri="sip:123456788@XXX.XXX.XX.XXX",
response="fd3b9b4d26c57a36ca4b1e96cca350ff", algorithm=MD5
Subject: no subject
User-Agent: XXXXX UserAgent/1.0
Max-Forwards: 70
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Type: application/SDP
Accept: application/SDP, text/plain
Accept-Encoding: identity
Content-Length: 287
v=0
o=XXXXXX 11375833482 11375833482 IN IP4 172.28.174.25
s=VaxSoft Inc.
c=IN IP4 172.28.174.25
t=0 0
m=audio 15000 RTP/AVP 3 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--- (16 headers 12 lines)---
Using INVITE request as basis request - 7469ea-26568-43ce24f4@172.28.174.25
Sending to 172.28.174.25 : 5060 (NAT)
Found user 'XXXXXX'
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.28.174.25:15000
Peer video RTP is at port 172.28.174.25:65535
Found description format GSM
Found description format iLBC
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x40e (gsm|ulaw|alaw|ilbc), peer - audio=0x40e
(gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e
(gsm|ulaw|alaw|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 123456788 in prepaid (domain XXX.XXX.XX.XXX)
list_route: hop: <sip:XXXXXX@172.28.174.25:5060>
Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 4 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:123456788@XXX.XXX.XX.XXX>
Content-Length: 0
---
We're at XXX.XXX.XX.XXX port 17532
Video is at XXX.XXX.XX.XXX port 14338
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 66.193.155.2:46478:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.28.174.25:5060;received=66.193.155.2
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>;tag=as2894778b
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 4 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:123456788@XXX.XXX.XX.XXX>
Content-Type: application/sdp
Content-Length: 291
TestServer*CLI>
v=0
o=root 4095 4095 IN IP4 XXX.XXX.XX.XXX
s=session
c=IN IP4 XXX.XXX.XX.XXX
t=0 0
m=audio 17532 RTP/AVP 98 3 0 8 101
a=rtpmap:98 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
ACK sip:123456788@XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX@XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788@XXX.XXX.XX.XXX>;tag=as2894778b
Call-ID: 7469ea-26568-43ce24f4@172.28.174.25
CSeq: 4 ACK
User-Agent: XXXXX UserAgent/1.0
Content-Length: 0
Below is the RTP Logs:
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37407, ts 878720, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5884, ts 897565, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37408, ts 878880, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5885, ts 897725, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37409, ts 879040, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37410, ts 879200, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5886, ts 897885, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5887, ts 898045, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37411, ts 879360, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5888, ts 898205, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37412, ts 879520, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5889, ts 898365, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37413, ts 879680, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5890, ts 898525, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37414, ts 879840, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5891, ts 898685, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37415, ts 880000, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5892, ts 898845, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37416, ts 880160, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5893, ts 899005, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37417, ts 880320, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5894, ts 899165, len 33)
Sent RTP packet to 66.193.155.2:15000 (type 3, seq 37418, ts 880480, len 33)
Got RTP packet from 66.193.155.2:15000 (type 3, seq 5895, ts 899325, len 33)
Regards,
Kengie
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20060118/d98e263a/attachment.htm