Does astersik supports tcp also

Asterisk Asterisk Support
1

I have install asterisk 1.4.19 on my linux system .Since asterisk supports UDP does it supports to the TCP also .I am facing a problem a voice delay when a call is initiated then for first 10 seconds iam not able t o hear the voice from my application(windows based) whereas the other user who is using xlite on sip media able to hear easily.My windows based application has a default settings of H323 i.e first priority is given to h323 that supports TCP .I think because of that only 10 seconds delay is there.

Thats why i wanted to know does asterisk supports tcp also

2

Asterisk 1.6 should support sip over tcp; please remember that 1.6 is still in beta.

Cheers.

Marco Bruni

3

While on the surface it may seem like a good idea to do your RTP (Real Time Protocol) over TCP, it’s really not.

TCP (Transmission Control Protocol) is designed to make sure that data gets to it’s destination. To do that it may delay sending data, resend data, and even send packets out of order knowing that the device on the other end will re-assemble them in the correct order. This behavior would be devastating to a real time audio stream. With TCP, getting the data there immediately is secondary to making sure it arrives perfectly.

UDP (User Datagram Protocol) has no such checking nor does it even care if a packet has made it to the distant end. It simply sends the next packet it has to send. The next packet goes as soon as it is ready. (And so on.)

This behavior is actually preferred. Why would you want to receive packets that should have been sent 1/2 or 1 second ago? By now, that audio has been heard, and new audio is being sent. You now want to hear the new audio. The old packets serve no purpose in the current audio stream.

It’s a lot like digital television. If a packet is lost, you see pixelation on the screen, and the image is distorted temporarily. When it recovers, would you want to see the packets that were lost earlier? Of course not. Those packets are not part of the current image. They’re useless.

TCP will likely not correct your problem, and may actually introduce new call quality issues.

4

Hi dufus
I think there is little bit of misunderstanding what i want to say is whenever i am calling from xlite to my sip application(vice versa) & call connects the first 10 seconds whatever a xlite client is saying to my application he is not able to hear. But whatever application client is saying to xlite client he is able to hear perfectly
After the first 10 seconds are over then there is a perfect communication between both the clients.

any suggestion

5

Without knowing more about your setup (softphone client, SIP/H323 setups, seeing debug screens, etc…) I really can’t offer any suggestions.

6

Hi dufus
As per your saying i gave debug command from my sip application I got

[code]<— SIP read from 192.168.1.77:1043 —>
INVITE sip:5000@192.168.1.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:8400
Max-Forwards: 70
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 1 INVITE
Contact: sip:192.168.1.77:8400
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 755

v=0
o=- 0 0 IN IP4 192.168.1.77
s=session
c=IN IP4 192.168.1.77
b=CT:1000
t=0 0
m=audio 53608 RTP/AVP 97 111 112 6 0 8 4 5 3 101
k=base64:KRhvkmsZ+fa5FGhpvWXKiaqCPV1bGeUcgFm/+G8bPrc
a=rtpmap:97 red/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:112 G7221/16000
a=fmtp:112 bitrate=24000
a=rtpmap:6 DVI4/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=encryption:optional
m=video 37786 RTP/AVP 34 31
k=base64:CKXqJj/itESd43xKs/1q5fq0PsZqLbpiVkhuIDKTklQ
a=recvonly
a=rtpmap:34 H263/90000
a=rtpmap:31 H261/90000
a=encryption:optional
m=application 1503 tcp msdata
a=sendonly
a=encryption:optional

<------------->
— (11 headers 31 lines) —
Sending to 192.168.1.77 : 8400 (no NAT)
Using INVITE request as basis request - 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
Found user '8888’
Found RTP audio format 97
Found RTP audio format 111
Found RTP audio format 112
Found RTP audio format 6
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 5
Found RTP audio format 3
Found RTP audio format 101
Found RTP video format 34
Found RTP video format 31
[Jul 31 09:56:21] WARNING[3151]: chan_sip.c:5140 process_sdp: Unsupported SDP media type in offer: application 1503 tcp msdata
Peer audio RTP is at port 192.168.1.77:53608
Found unknown media description format red for ID 97
Found unknown media description format SIREN for ID 111
Found unknown media description format G7221 for ID 112
Found audio description format DVI4 for ID 6
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format DVI4 for ID 5
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Found unknown media description format H263 for ID 34
Found unknown media description format H261 for ID 31
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2f (g723|gsm|ulaw|alaw|adpcm)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.77:53608
Looking for 5000 in sip (domain 192.168.1.46)
list_route: hop: sip:192.168.1.77:8400

<— Transmitting (no NAT) to 192.168.1.77:8400 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.77:8400;received=192.168.1.77
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5000@192.168.1.46
Content-Length: 0

<------------>
– Executing [5000@sip:1] Dial(“SIP/8888-093e8bb8”, “SIP/5000|10”) in new stack
– Called 5000
– SIP/5000-093ed640 is ringing

<— Transmitting (no NAT) to 192.168.1.77:8400 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.77:8400;received=192.168.1.77
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46;tag=as0893cea8
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5000@192.168.1.46
Content-Length: 0

<------------>
Audio is at 192.168.1.46 port 14890
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

<— Transmitting (no NAT) to 192.168.1.77:8400 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.77:8400;received=192.168.1.77
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46;tag=as0893cea8
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5000@192.168.1.46
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 3146 3146 IN IP4 192.168.1.46
s=session
c=IN IP4 192.168.1.46
t=0 0
m=audio 14890 RTP/AVP 3 0 8 101
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 - - - -
a=ptime:20
a=sendrecv

<------------>
– SIP/5000-093ed640 answered SIP/8888-093e8bb8
Audio is at 192.168.1.46 port 14890
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 (no NAT) to 192.168.1.77:8400 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.77:8400;received=192.168.1.77
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46;tag=as0893cea8
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5000@192.168.1.46
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 3146 3147 IN IP4 192.168.1.46
s=session
c=IN IP4 192.168.1.46
t=0 0
m=audio 14890 RTP/AVP 3 0 8 101
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 - - - -
a=ptime:20
a=sendrecv

<------------>
– Native bridging SIP/8888-093e8bb8 and SIP/5000-093ed640

<— SIP read from 192.168.1.77:1043 —>
ACK sip:5000@192.168.1.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:8400
Max-Forwards: 70
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46;tag=as0893cea8
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 1 ACK
User-Agent: RTC/1.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —
set_destination: Parsing sip:192.168.1.77:8400 for address/port to send to
set_destination: set destination to 192.168.1.77, port 8400
Audio is at 192.168.1.46 port 14890
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.77:8400:
INVITE sip:192.168.1.77:8400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK06144c83
From: sip:5000@192.168.1.46;tag=as0893cea8
To: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
Contact: sip:5000@192.168.1.46
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 239

v=0
o=root 3146 3148 IN IP4 192.168.1.130
s=session
c=IN IP4 192.168.1.130
t=0 0
m=audio 5063 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from 192.168.1.77:1043 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK06144c83
From: sip:5000@192.168.1.46;tag=as0893cea8
To: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 102 INVITE
User-Agent: RTC/1.2
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from 192.168.1.77:1043 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK06144c83
From: sip:5000@192.168.1.46;tag=as0893cea8
To: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 102 INVITE
Contact: sip:192.168.1.77:8400
User-Agent: RTC/1.2
Content-Type: application/sdp
Content-Length: 239

v=0
o=- 0 0 IN IP4 192.168.1.77
s=session
c=IN IP4 192.168.1.77
b=CT:1000
t=0 0
m=audio 53608 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 0 RTP/AVP 34
m=application 0 tcp msdata

<------------->
— (10 headers 12 lines) —
Found RTP audio format 8
Found RTP audio format 101
Found RTP video format 34
[Jul 31 09:56:23] WARNING[3151]: chan_sip.c:5140 process_sdp: Unsupported SDP media type in offer: application 0 tcp msdata
Peer audio RTP is at port 192.168.1.77:53608
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x80008 (alaw|h263)/video=0x80000 (h263), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.77:53608
set_destination: Parsing sip:192.168.1.77:8400 for address/port to send to
set_destination: set destination to 192.168.1.77, port 8400
Transmitting (no NAT) to 192.168.1.77:8400:
ACK sip:192.168.1.77:8400 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK2f5df53d
From: sip:5000@192.168.1.46;tag=as0893cea8
To: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
Contact: sip:5000@192.168.1.46
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<— SIP read from 192.168.1.77:1043 —>
BYE sip:5000@192.168.1.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.77:8400
Max-Forwards: 70
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46;tag=as0893cea8
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 2 BYE
User-Agent: RTC/1.2
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.77 : 8400 (no NAT)

<— Transmitting (no NAT) to 192.168.1.77:8400 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.77:8400;received=192.168.1.77
From: “192.168.1.77” sip:8888@192.168.1.46;tag=11c2d3f5946e408aa68d3ce2540aecfd;epid=faab85e73a
To: sip:5000@192.168.1.46;tag=as0893cea8
Call-ID: 89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:5000@192.168.1.46
Content-Length: 0

<------------>
== Spawn extension (sip, 5000, 1) exited non-zero on 'SIP/8888-093e8bb8’
Really destroying SIP dialog ‘89d9715b93dc4aaeb0143fe4ca493139@192.168.1.77’ Method: BYE

*CLI> sip set debug ip 192.168.1.77Reliably Transmitting (no NAT) to 192.168.1.77:5060:
OPTIONS sip:192.168.1.77 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK74579356
From: “asterisk” sip:asterisk@192.168.1.46;tag=as6ea078ad
To: sip:192.168.1.77
Contact: sip:asterisk@192.168.1.46
Call-ID: 14aba4f83b092e0574a54a44327b1ab4@192.168.1.46
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 31 Jul 2008 04:27:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

<— SIP read from 192.168.1.77:1043 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK74579356
From: “asterisk” sip:asterisk@192.168.1.46;tag=as6ea078ad
To: sip:192.168.1.77;tag=eb060a96ae11499783352e0025a1c567
Call-ID: 14aba4f83b092e0574a54a44327b1ab4@192.168.1.46
CSeq: 102 OPTIONS
User-Agent: RTC/1.2
Allow: INVITE, BYE, OPTIONS, MESSAGE, ACK, CANCEL, NOTIFY, SUBSCRIBE, INFO, REFER
Content-Type: application/sdp
Content-Length: 545[/code]

When i called from xlite to my application softphone .I got this

[code]
<------------>
– Executing [8888@sip:1] Dial(“SIP/5000-093eb988”, “SIP/8888”) in new stack
– Called 8888
– SIP/8888-093f1ef0 is ringing

<— Transmitting (no NAT) to 192.168.1.130:5061 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.130:5061;branch=z9hG4bK-d87543-333a7f63a3146f0c-1–d87543-;rport;received=192.168.1.130
From: "5000"sip:5000@192.168.1.46;tag=127acb11
To: "8888"sip:8888@192.168.1.46;tag=as29d3555b
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8888@192.168.1.46
Content-Length: 0

<------------>
– SIP/8888-093f1ef0 answered SIP/5000-093eb988
Audio is at 192.168.1.46 port 17720
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.130:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.130:5061;branch=z9hG4bK-d87543-333a7f63a3146f0c-1–d87543-;rport;received=192.168.1.130
From: "5000"sip:5000@192.168.1.46;tag=127acb11
To: "8888"sip:8888@192.168.1.46;tag=as29d3555b
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8888@192.168.1.46
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 3146 3146 IN IP4 192.168.1.46
s=session
c=IN IP4 192.168.1.46
t=0 0
m=audio 17720 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Native bridging SIP/5000-093eb988 and SIP/8888-093f1ef0

<— SIP read from 192.168.1.130:5061 —>
ACK sip:8888@192.168.1.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5061;branch=z9hG4bK-d87543-b965f463635a4b64-1–d87543-;rport
Max-Forwards: 70
Contact: sip:5000@192.168.1.130:5061
To: "8888"sip:8888@192.168.1.46;tag=as29d3555b
From: “5000"sip:5000@192.168.1.46;tag=127acb11
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 2 ACK
Proxy-Authorization: Digest username=“5000”,realm=“asterisk”,nonce=“2cd22d65”,uri="sip:8888@192.168.1.46”,response=“ff3128e9f07716955b0eb901e25f8004”,algorithm=MD5
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 0

<------------->
— (11 headers 0 lines) —
set_destination: Parsing sip:5000@192.168.1.130:5061 for address/port to send to
set_destination: set destination to 192.168.1.130, port 5061
Audio is at 192.168.1.46 port 17720
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.130:5061:
INVITE sip:5000@192.168.1.130:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK62680f64
From: "8888"sip:8888@192.168.1.46;tag=as29d3555b
To: "5000"sip:5000@192.168.1.46;tag=127acb11
Contact: sip:8888@192.168.1.46
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 3146 3147 IN IP4 192.168.1.77
s=session
c=IN IP4 192.168.1.77
t=0 0
m=audio 21954 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from 192.168.1.130:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK62680f64
Contact: sip:5000@192.168.1.130:5061
To: "5000"sip:5000@192.168.1.46;tag=127acb11
From: "8888"sip:8888@192.168.1.46;tag=as29d3555b
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1002tx stamp 29712
Content-Length: 240

v=0
o=- 9 3 IN IP4 192.168.1.130
s=<CounterPath eyeBeam 1.5>
c=IN IP4 192.168.1.130
t=0 0
m=audio 5063 RTP/AVP 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:2F132F94D44546168452B06435CA8106

<------------->
— (11 headers 10 lines) —
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.130:5063
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.130:5063
set_destination: Parsing sip:5000@192.168.1.130:5061 for address/port to send to
set_destination: set destination to 192.168.1.130, port 5061
Transmitting (no NAT) to 192.168.1.130:5061:
ACK sip:5000@192.168.1.130:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK4a378ac2
From: "8888"sip:8888@192.168.1.46;tag=as29d3555b
To: "5000"sip:5000@192.168.1.46;tag=127acb11
Contact: sip:8888@192.168.1.46
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

<— SIP read from 192.168.1.130:5061 —>
BYE sip:8888@192.168.1.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.130:5061;branch=z9hG4bK-d87543-8f5aef02663a686c-1–d87543-;rport
Max-Forwards: 70
Contact: sip:5000@192.168.1.130:5061
To: "8888"sip:8888@192.168.1.46;tag=as29d3555b
From: “5000"sip:5000@192.168.1.46;tag=127acb11
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 3 BYE
Proxy-Authorization: Digest username=“5000”,realm=“asterisk”,nonce=“2cd22d65”,uri="sip:8888@192.168.1.46”,response=“e45f46514bf153f910f16f6889b3f8f6”,algorithm=MD5
User-Agent: X-Lite release 1002tx stamp 29712
Reason: SIP;description="User Hung Up"
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.1.130 : 5061 (NAT)

<— Transmitting (NAT) to 192.168.1.130:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.130:5061;branch=z9hG4bK-d87543-8f5aef02663a686c-1–d87543-;rport;received=192.168.1.130
From: "5000"sip:5000@192.168.1.46;tag=127acb11
To: "8888"sip:8888@192.168.1.46;tag=as29d3555b
Call-ID: 1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8888@192.168.1.46
Content-Length: 0

<------------>
== Spawn extension (sip, 8888, 1) exited non-zero on 'SIP/5000-093eb988’
Really destroying SIP dialog ‘1725a8230655e875NzYxZThjNTBiZDgxZGVjM2QxMzMwNGFkNTg3YzczN2Q.’ Method: BYE

<— SIP read from 192.168.1.130:5061 —>[/code]