[Asterisk-Users] Re: SIP RTP Negotiation

Kenige Ho Tue, 24 Jan 2006 20:10:40 -0800

Hi Asterisk-Users,

Please help as it is out of my league and understand as why the call would be silent or partly silent (calling party can't hear the called party, but called party can hear). Thanks in advance.

Regards,

Kengie

On 1/19/06, Kenige Ho <[EMAIL PROTECTED]> wrote:

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:[EMAIL PROTECTED]>
To: XXXXXX <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
Contact: "XXXXXX" <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>
To: XXXXXX <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: < sip:[EMAIL PROTECTED] >
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:[EMAIL PROTECTED]>
To: XXXXXX <sip:[EMAIL PROTECTED]>;tag=as504de7b8
Call-ID: [EMAIL PROTECTED]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
WWW-Authenticate: Digest realm="asterisk", nonce="31e7aa76"
Content-Length: 0

---
Scheduling destruction of call '[EMAIL PROTECTED]' 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:[EMAIL PROTECTED] >
To: XXXXXX <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
Contact: "XXXXXX" <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>
To: XXXXXX <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: < sip:[EMAIL PROTECTED] >
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:[EMAIL PROTECTED]>
To: XXXXXX <sip:[EMAIL PROTECTED]>;tag=as504de7b8
Call-ID: [EMAIL PROTECTED]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Expires: 3600
Contact: <sip:[EMAIL PROTECTED]:5060>;expires=3600
Date: Wed, 18 Jan 2006 11:20:15 GMT
Content-Length: 0

---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Destroying call '[EMAIL PROTECTED]'
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 '[EMAIL PROTECTED]'
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX < sip:[EMAIL PROTECTED] >;tag=174cf52
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 3 INVITE
Contact: <sip:[EMAIL PROTECTED] :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 - [EMAIL PROTECTED]
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:[EMAIL PROTECTED]>;tag=174cf52
To: < sip:[EMAIL PROTECTED]>;tag=as733e9adc
Call-ID: [EMAIL PROTECTED]
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: < sip:[EMAIL PROTECTED]>
Proxy-Authenticate: Digest realm="asterisk", nonce="452293c4"
Content-Length: 0

---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Found user 'XXXXXX'
TestServer*CLI>
<-- SIP read from 66.193.155.2:46478:
ACK sip:[EMAIL PROTECTED] :5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX < sip:[EMAIL PROTECTED]>;tag=174cf52
To: <sip:[EMAIL PROTECTED]>;tag=as733e9adc
Call-ID: [EMAIL PROTECTED]
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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX < sip:[EMAIL PROTECTED]>;tag=174cf52
To: <sip:[EMAIL PROTECTED] >
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INVITE
Contact: <sip:[EMAIL PROTECTED] :5060>
Proxy-Authorization: Digest username="XXXXXX", realm="asterisk", nonce="452293c4", uri=" sip:[EMAIL PROTECTED]", 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 - [EMAIL PROTECTED]
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:[EMAIL PROTECTED] :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:[EMAIL PROTECTED] >;tag=174cf52
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED] >
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:[EMAIL PROTECTED]>;tag=174cf52
To: < sip:[EMAIL PROTECTED]>;tag=as2894778b
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: < sip:[EMAIL PROTECTED]>
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:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 172.28.174.25:5060
From: XXXXXX < sip:[EMAIL PROTECTED]>;tag=174cf52
To: <sip:[EMAIL PROTECTED]>;tag=as2894778b
Call-ID: [EMAIL PROTECTED]
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)

Regards,

Kengie

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[Asterisk-Users] Re: SIP RTP Negotiation

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