[Asterisk-Users] SIP RTP Negotiation

Wed Jan 18 23:45:37 MST 2006

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 at XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
CSeq: 1 REGISTER
Contact: "XXXXXX" <sip:XXXXXX at 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 at XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec at 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 at 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 at XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=as504de7b8
Call-ID: 74494a-1654e-43ce24ec at 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 at XXX.XXX.XX.XXX>
WWW-Authenticate: Digest realm="asterisk", nonce="31e7aa76"
Content-Length: 0

---
Scheduling destruction of call '74494a-1654e-43ce24ec at 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 at XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec at XXX.XXX.XX.XXX
CSeq: 2 REGISTER
Contact: "XXXXXX" <sip:XXXXXX at 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 at XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>
Call-ID: 74494a-1654e-43ce24ec at 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 at 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 at XXX.XXX.XX.XXX>
To: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=as504de7b8
Call-ID: 74494a-1654e-43ce24ec at 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 at 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 at XXX.XXX.XX.XXX' in
15000 ms
Destroying call '4CEFE29BD3864FFFA788C2D43F4E6FBD at 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 at 192.168.1.35'
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
INVITE sip:123456788 at XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>
Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
CSeq: 3 INVITE
Contact: <sip:XXXXXX at 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 at 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 at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>;tag=as733e9adc
Call-ID: 7469ea-26568-43ce24f4 at 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 at XXX.XXX.XX.XXX>
Proxy-Authenticate: Digest realm="asterisk", nonce="452293c4"
Content-Length: 0

---
Scheduling destruction of call '7469ea-26568-43ce24f4 at 172.28.174.25' in
15000 ms
Found user 'XXXXXX'
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
ACK sip:123456788 at XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>;tag=as733e9adc
Call-ID: 7469ea-26568-43ce24f4 at 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 at XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>
Call-ID: 7469ea-26568-43ce24f4 at 172.28.174.25
CSeq: 4 INVITE
Contact: <sip:XXXXXX at 172.28.174.25:5060>
Proxy-Authorization: Digest username="XXXXXX", realm="asterisk",
nonce="452293c4", uri="sip:123456788 at 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 at 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 at 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 at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>
Call-ID: 7469ea-26568-43ce24f4 at 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 at 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 at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>;tag=as2894778b
Call-ID: 7469ea-26568-43ce24f4 at 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 at 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 at XXX.XXX.XX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX <sip:XXXXXX at XXX.XXX.XX.XXX>;tag=174cf52
To: <sip:123456788 at XXX.XXX.XX.XXX>;tag=as2894778b
Call-ID: 7469ea-26568-43ce24f4 at 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
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