正常的codec协商包
<--- SIP read from UDP:192.168.4.18:5067--->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK424774178
From: "300"<sip:[email protected]>;tag=778739804
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5067>
Authorization: Digestusername="300", realm="asterisk",nonce="61bc3751", uri="sip:[email protected]",response="f9668313c2a854c6b5eb670899c79767",algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.61.0.70
Supported: replaces
Allow-Events:talk,hold,conference,refer,check-sync
Content-Length: 282
v=0
o=- 21051 21051 IN IP4 192.168.4.18
s=SDP data
c=IN IP4 192.168.4.18
t=0 0
m=audio 11782 RTP/AVP 18 8 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 14 lines) ---
Sending to 192.168.4.18 : 5067 (no NAT)
Using INVITE request as basis request [email protected]
Found peer '300' for '300' from192.168.4.18:5067
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G729 for ID18
Found audio description format PCMA for ID8
Found audio description format PCMU for ID0
Found audio description formattelephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0(nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1(telephone-event), peer - 0x1 (telephone-event), combined - 0x1(telephone-event)
Peer audio RTP is at port192.168.4.18:11782
Looking for 301 in DLPN_DialPlan300 (domain 192.168.4.122)
list_route: hop:<sip:[email protected]:5067>
<--- Transmitting (no NAT) to192.168.4.18:5067 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK424774178;received=192.168.4.18
From: "300"<sip:[email protected]>;tag=778739804
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
-- Executing [301@DLPN_DialPlan300:1]Macro("SIP/300-00000002", "stdexten,301,SIP/301") in newstack
-- Executing [s@macro-stdexten:1] Set("SIP/300-00000002","IsFromOutside=0") in new stack
-- Executing [s@macro-stdexten:2] Macro("SIP/300-00000002","realstexten,301,SIP/301,tTkKWwXx") in new stack
-- Executing [s@macro-realstexten:1] Set("SIP/300-00000002","DYNAMIC_FEATURES=twstart") in new stack
-- Executing [s@macro-realstexten:2]GotoIf("SIP/300-00000002", "0?Blacklist-Handle,s,1") in newstack
-- Executing [s@macro-realstexten:3] Set("SIP/300-00000002","TIMEOUT(absolute)=6000") in new stack
Channel will hangup at 2012-10-0516:54:01.274 ???.
-- Executing [s@macro-realstexten:4] Set("SIP/300-00000002","CKTSETTRANSFER=0") in new stack
-- Executing [s@macro-realstexten:5] Set("SIP/300-00000002","REALARG1=301") in new stack
-- Executing [s@macro-realstexten:6]GotoIf("SIP/300-00000002", "0>0?follow-me,1") in newstack
-- Executing [s@macro-realstexten:7]GotoIf("SIP/300-00000002", "0>0?vm-u,1") in new stack
-- Executing [s@macro-realstexten:8] Set("SIP/300-00000002","RINGTIME=30") in new stack
-- Executing [s@macro-realstexten:9]CktStdCall("SIP/300-00000002", "srtpfor,SIP/301,novalue")in new stack
-- Executing [s@macro-realstexten:10] Set("SIP/300-00000002","_SIPSRTP=0") in new stack
-- Executing [s@macro-realstexten:11] Dial("SIP/300-00000002","SIP/301,30,tTkKWwXx") in new stack
==Using SIP RTP TOS bits 184
==Using SIP RTP CoS mark 5
==Using SIP VRTP TOS bits 136
==Using SIP VRTP CoS mark 4
Audio is at 192.168.4.122 port 10282
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
<--- SIP read from UDP:192.168.4.18:5067--->
SIP/2.0 200 OK
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK7e70b194;rport
From: <sip:[email protected]>;tag=as5a4d8aee
To: "300"<sip:[email protected]>;tag=43927923
Call-ID: [email protected]
CSeq: 102 NOTIFY
User-Agent: Yealink SIP-T28P 2.61.0.70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
SIP Response message for INCOMING dialogNOTIFY arrived
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) toSDP
Reliably Transmitting (no NAT) to192.168.4.26:5066:
INVITE sip:[email protected]:5066 SIP/2.0
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK7ca8a118;rport
Max-Forwards: 70
From: "300"<sip:[email protected]>;tag=as09e7c96d
To: <sip:[email protected]:5066>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: MyPBX
Date: Fri, 05 Oct 2012 07:14:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 567018984 567018984 IN IP4 192.168.4.122
s=Asterisk PBX SVN--r2721M
c=IN IP4 192.168.4.122
t=0 0
m=audio 10282 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
-- Called 301
<--- SIP read from UDP:192.168.4.26:5066--->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK53008412;rport
From: "300"<sip:[email protected]>;tag=as20c8c805
To: <sip:[email protected]:5066>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Yealink SIP-T28P 2.61.0.80
Content-Length: 0
<--- SIP read from UDP:192.168.4.26:5066--->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK53008412;rport
From: "300"<sip:[email protected]>;tag=as20c8c805
To:<sip:[email protected]:5066>;tag=1545051705
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5066>
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T28P 2.61.0.80
Allow-Events:talk,hold,conference,refer,check-sync
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
-- SIP/301-00000005 is ringing
<--- Transmitting (no NAT) to192.168.4.18:5067 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK948972928;received=192.168.4.18
From: "300"<sip:[email protected]>;tag=397490258
To: <sip:[email protected]>;tag=as64a49174
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0
<--- SIP read from UDP:192.168.4.26:5066--->
SIP/2.0 200 OK
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK53008412;rport
From: "300"<sip:[email protected]>;tag=as20c8c805
To: <sip:[email protected]:5066>;tag=1545051705
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5066>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T28P 2.61.0.80
Content-Length: 234
v=0
o=- 20286 20286 IN IP4 192.168.4.26
s=SDP data
c=IN IP4 192.168.4.26
t=0 0
m=audio 11780 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
a=ptime:20
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
--- (11 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID18
Found audio description formattelephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x100(g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities (dtmf): us - 0x1(telephone-event), peer - 0x1 (telephone-event), combined - 0x1(telephone-event)
Peer audio RTP is at port 192.168.4.26:11780
-- g729tolin_new
-- use count: 1
-- lintog729_new
-- use count: 2
list_route: hop:<sip:[email protected]:5066>
set_destination: Parsing<sip:[email protected]:5066> for address/port to send to
set_destination: set destination to192.168.4.26, port 5066
Transmitting (no NAT) to 192.168.4.26:5066:
ACK sip:[email protected]:5066 SIP/2.0
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK66d1d8f7;rport
Max-Forwards: 70
From: "300"<sip:[email protected]>;tag=as20c8c805
To: <sip:[email protected]:5066>;tag=1545051705
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: MyPBX
Content-Length: 0
不支持发起方codec
<--- SIP read from UDP:192.168.4.18:5067--->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK313013269
From: "300"<sip:[email protected]>;tag=296948853
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5067>
Authorization: Digestusername="300", realm="asterisk", nonce="387c28fd",uri="sip:[email protected]", response="d6bec2f58aceb8cf9183efdf75ed7301",algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.61.0.70
Supported: replaces
Allow-Events:talk,hold,conference,refer,check-sync
Content-Length: 235
v=0
o=- 21047 21047 IN IP4 192.168.4.18
s=SDP data
c=IN IP4 192.168.4.18
t=0 0
m=audio 11780 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.4.18 : 5067 (no NAT)
Using INVITE request as basis request [email protected]
Found peer '300' for '300' from192.168.4.18:5067
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID8
Found audio description format PCMU for ID0
Found audio description formattelephone-event for ID 101
Capabilities: us - 0x100 (g729), peer -audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0(nothing)
Non-codec capabilities (dtmf): us - 0x1(telephone-event), peer - 0x1 (telephone-event), combined - 0x1(telephone-event)
[2012-10-05 14:47:38] NOTICE[765]:chan_sip.c:8819 process_sdp: No compatible codecs, not accepting this offer!
<--- Reliably Transmitting (no NAT) to192.168.4.18:5067 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK313013269;received=192.168.4.18
From: "300"<sip:[email protected]>;tag=296948853
To:<sip:[email protected]>;tag=as7bd77b6d
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
不支持被叫方codec
<--- SIP read from UDP:192.168.4.18:5067--->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK1575142430
From: "300"<sip:[email protected]>;tag=920581371
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5067>
Authorization: Digestusername="300", realm="asterisk", nonce="15ab2d9a",uri="sip:[email protected]", response="a5800d22176648a208ac4c70a04e490b", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.61.0.70
Supported: replaces
Allow-Events:talk,hold,conference,refer,check-sync
Content-Length: 306
v=0
o=- 21046 21046 IN IP4 192.168.4.18
s=SDP data
c=IN IP4 192.168.4.18
t=0 0
m=audio 11780 RTP/AVP 8 0 9 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.4.18 : 5067 (no NAT)
Using INVITE request as basis request [email protected]
Found peer '300' for '300' from192.168.4.18:5067
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID8
Found audio description format PCMU for ID0
Found audio description format G722 for ID9
Found audio description format G729 for ID18
Found audio description formattelephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer -audio=0x110c(ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc(ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1(telephone-event), peer - 0x1 (telephone-event), combined - 0x1(telephone-event)
Peer audio RTP is at port192.168.4.18:11780
Looking for 301 in DLPN_DialPlan300 (domain 192.168.4.122)
list_route: hop:<sip:[email protected]:5067>
<--- Transmitting (no NAT) to192.168.4.18:5067 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK1575142430;received=192.168.4.18
From: "300"<sip:[email protected]>;tag=920581371
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0
<------------>
-- Executing [301@DLPN_DialPlan300:1] Macro("SIP/300-00000019","stdexten,301,SIP/301") in new stack
-- Executing [s@macro-stdexten:1] Set("SIP/300-00000019","IsFromOutside=0") in new stack
-- Executing [s@macro-stdexten:2] Macro("SIP/300-00000019","realstexten,301,SIP/301,tTkKWwXx") in new stack
-- Executing [s@macro-realstexten:1] Set("SIP/300-00000019","DYNAMIC_FEATURES=twstart") in new stack
-- Executing [s@macro-realstexten:2]GotoIf("SIP/300-00000019", "0?Blacklist-Handle,s,1") in newstack
-- Executing [s@macro-realstexten:3] Set("SIP/300-00000019","TIMEOUT(absolute)=6000") in new stack
Channel will hangup at 2012-10-0516:19:21.493 ???.
-- Executing [s@macro-realstexten:4] Set("SIP/300-00000019","CKTSETTRANSFER=0") in new stack
-- Executing [s@macro-realstexten:5] Set("SIP/300-00000019","REALARG1=301") in new stack
-- Executing [s@macro-realstexten:6]GotoIf("SIP/300-00000019", "0>0?follow-me,1") in newstack
-- Executing [s@macro-realstexten:7]GotoIf("SIP/300-00000019", "0>0?vm-u,1") in new stack
--Executing [s@macro-realstexten:8] Set("SIP/300-00000019","RINGTIME=30") in new stack
-- Executing [s@macro-realstexten:9]CktStdCall("SIP/300-00000019", "srtpfor,SIP/301,novalue")in new stack
-- Executing [s@macro-realstexten:10] Set("SIP/300-00000019","_SIPSRTP=0") in new stack
-- Executing [s@macro-realstexten:11] Dial("SIP/300-00000019","SIP/301,30,tTkKWwXx") in new stack
==Using SIP RTP TOS bits 184
==Using SIP RTP CoS mark 5
==Using SIP VRTP TOS bits 136
==Using SIP VRTP CoS mark 4
Audio is at 192.168.4.122 port 11150
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x1000 (g722) to SDP
Adding non-codec 0x1 (telephone-event) toSDP
Reliably Transmitting (no NAT) to192.168.4.26:5067:
INVITE sip:[email protected]:5067 SIP/2.0
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK516d07fb;rport
Max-Forwards: 70
From: "300"<sip:[email protected]>;tag=as187de9c6
To: <sip:[email protected]:5067>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: MyPBX
Date: Sun, 02 Jan 2000 21:36:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 1762416231 1762416231 IN IP4192.168.4.122
s=Asterisk PBX SVN--r2721M
c=IN IP4 192.168.4.122
t=0 0
m=audio 11150 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 301
<--- SIP read from UDP:192.168.4.26:5067--->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK516d07fb;rport
From: "300"<sip:[email protected]>;tag=as187de9c6
To: <sip:[email protected]:5067>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Yealink SIP-T28P 2.61.0.80
Content-Length: 0
<--- SIP read from UDP:192.168.4.26:5067--->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK516d07fb;rport
From: "300"<sip:[email protected]>;tag=as187de9c6
To:<sip:[email protected]:5067>;tag=1908231474
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Yealink SIP-T28P 2.61.0.80
Warning: 304 info "No supported mediatype found"
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.4.26:5067:
ACK sip:[email protected]:5067 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.122:5060;branch=z9hG4bK516d07fb;rport
Max-Forwards: 70
From: "300"<sip:[email protected]>;tag=as187de9c6
To:<sip:[email protected]:5067>;tag=1908231474
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: MyPBX
Content-Length: 0
==Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-realstexten:12] Goto("SIP/300-00000019","s-CONGESTION,1") in new stack
-- Goto (macro-realstexten,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-realstexten:1]GotoIf("SIP/300-00000019", "0>0?cktcheck,1") in newstack
-- Executing [s-CONGESTION@macro-realstexten:2]Goto("SIP/300-00000019", "s-NOANSWER,1") in new stack
-- Goto (macro-realstexten,s-NOANSWER,1)
--Executing [s-NOANSWER@macro-realstexten:1] GotoIf("SIP/300-00000019","1>0?follow-me,1:1-out,1") in new stack
-- Goto (macro-realstexten,follow-me,1)
-- Executing [follow-me@macro-realstexten:1]GotoIf("SIP/300-00000019", "1>0?vm-u,1") in new stack
-- Goto (macro-realstexten,vm-u,1)
-- Executing [vm-u@macro-realstexten:1]GotoIf("SIP/300-00000019", "0?vm-u,2:vm-unavail,1") in newstack
-- Goto (macro-realstexten,vm-unavail,1)
-- Executing [vm-unavail@macro-realstexten:1] VoiceMail("SIP/300-00000019","301,u") in new stack
Audio is at 192.168.4.122 port 10002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) toSDP
MyPBX*CLI>
<--- Reliably Transmitting (no NAT) to192.168.4.18:5067 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK1575142430;received=192.168.4.18
From: "300"<sip:[email protected]>;tag=920581371
To:<sip:[email protected]>;tag=as164cf809
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 264
MyPBX*CLI>
v=0
o=root 371020670 371020670 IN IP4 192.168.4.122
s=Asterisk PBX SVN--r2721M
c=IN IP4 192.168.4.122
t=0 0
m=audio 10002 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
分机互打协商过程解析
步骤1: MyPBX收到A的SDP后,协商确定A与MyPBX之间的最优编码.
步骤2: MyPBX发送SDP给B时,会把步骤-1协商好的最优编码放置于SDP编码的首位.B根据本地设置,确定B与MyPBX之间的编码.
设置codec的相关函数：
sip_new:
1: 为sip分机创建一个通道.
/*Set the native formats for audio andmerge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) |video | text;
2: 为该通道选择最优编码.
/*Set the native formats for audio andmerge in video */
fmt = ast_best_codec(tmp->nativeformats);
3: 赋值相关成员变量.
tmp->writeformat = fmt;
tmp->rawwriteformat = fmt;
tmp->readformat = fmt;
tmp->rawreadformat = fmt;
process_sdp:
1. 根据sdp包含的编码信息,设置通道的读写编码格式.
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
2. 在set_format函数中,根据原始和目标编码的类型,设置编码转换回调结构体.struct ast_trans_pvt;
步骤1 过程介绍
MyPBX收到A的SDP后,协商确定A与MyPBX之间的最优编码.
若A与MyPBX之间没有相同的编码支持,MyPBX返回488告知A.呼叫失败.
若A与MyPBX时间有多个相同的编码支持,MyPBX选择最优编码.发SDP包告知A.
编码的优化顺序一般为:ulaw|alaw|g729|g722
首选ulaw/alaw,编解码快,压缩比2:1.
次选g729,没有压缩,但也没有编解码的性能消耗.
再选g722,压缩比高,但编解码性能消耗较多.
<--- SIP read from UDP:192.168.4.18:5062--->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP192.168.4.18:5062;branch=z9hG4bK330291957
From: "500"<sip:[email protected]>;tag=1071929271
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.61.0.70
Supported: replaces
Allow-Events:talk,hold,conference,refer,check-sync
Content-Length: 306
v=0
o=- 21137 21137 IN IP4 192.168.4.18
s=SDP data
c=IN IP4 192.168.4.18
t=0 0
m=audio 11784 RTP/AVP 9 18 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 15 lines) ---
==Using SIP RTP TOS bits 184
==Using SIP RTP CoS mark 5
==Using SIP VRTP TOS bits 136
==Using SIP VRTP CoS mark 4
Sending to 192.168.4.18 : 5062 (NAT)
Using INVITE request as basis request [email protected]
Found peer '500' for '500' from 192.168.4.18:5062
<--- Reliably Transmitting (no NAT) to192.168.4.18:5062 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP192.168.4.18:5062;branch=z9hG4bK330291957;received=192.168.4.18
From: "500"<sip:[email protected]>;tag=1071929271
To: <sip:[email protected]>;tag=as6fbd5d05
Call-ID: [email protected]
CSeq: 1 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5,realm="asterisk", nonce="67ce7ca2"
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 192.168.4.18 : 5062 (no NAT)
Found peer '500' for '500' from192.168.4.18:5062
Looking for 505 in hintcontext (domain 192.168.5.44)
Scheduling destruction of SIP dialog'[email protected]' in 70000 ms (Method: SUBSCRIBE)
<--- SIP read from UDP:192.168.4.18:5062--->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP192.168.4.18:5062;branch=z9hG4bK1993271819
From: "500"<sip:[email protected]>;tag=1071929271
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5062>
Authorization: Digestusername="500", realm="asterisk",nonce="67ce7ca2", uri="sip:[email protected]",response="98b37c3e10bb52a4fc7be44746a6689e", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T28P 2.61.0.70
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 306
v=0
o=- 21137 21137 IN IP4 192.168.4.18
s=SDP data
c=IN IP4 192.168.4.18
t=0 0
m=audio 11784 RTP/AVP 9 18 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.4.18 : 5062 (no NAT)
Using INVITE request as basis request [email protected]
Found peer '500' for '500' from192.168.4.18:5062
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format G722 for ID9
Found audio description format G729 for ID18
Found audio description format PCMA for ID8
Found audio description format PCMU for ID0
Found audio description formattelephone-event for ID 101
[2012-10-08 01:20:41] NOTICE[549]:chan_sip.c:8805 process_sdp: p(17279596)->owner(86299164) Capabilities:
[2012-10-08 01:20:41] NOTICE[549]:chan_sip.c:8806 process_sdp: us -0x110c (ulaw|alaw|g729|g722)
[2012-10-08 01:20:41] NOTICE[549]:chan_sip.c:8807 process_sdp: peer -0x110c (ulaw|alaw|g729|g722)
[2012-10-08 01:20:41] NOTICE[549]: chan_sip.c:8808process_sdp: combined - 0x110c(ulaw|alaw|g729|g722)
Non-codec capabilities (dtmf): us - 0x1(telephone-event), peer - 0x1 (telephone-event), combined - 0x1(telephone-event)
Peer audio RTP is at port192.168.4.18:11784
Looking for 505 in DLPN_DialPlan500 (domain 192.168.5.44)
[2012-10-08 01:20:41] NOTICE[549]:chan_sip.c:6964 sip_new: sip_new set owner to chan(68177924 SIP/500-00000014)
list_route: hop:<sip:[email protected]:5062>
<--- Transmitting (no NAT) to192.168.4.18:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.18:5062;branch=z9hG4bK1993271819;received=192.168.4.18
From: "500"<sip:[email protected]>;tag=1071929271
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0
步骤2 过程介绍
MyPBX发送SDP给B时,会把步骤-1协商好的最优编码放置于SDP编码的首位.
若B不支持SDP包中的任一编码,B返回488告知MyPBX.
若B支持SDP包中的任一编码,选择最优编码,返回SDP告知MyPBX.(yealink话机返回的SDP信息中,只包括一个最优编码,而不是支持的所有编码)
[2012-10-08 01:20:41] NOTICE[1755]:chan_sip.c:6964 sip_new: sip_new set owner to chan(16494596 SIP/505-00000015)
Audio is at 192.168.5.44 port 11122
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) toSDP
Reliably Transmitting (no NAT) to192.168.4.135:5064:
INVITE sip:[email protected]:5064 SIP/2.0
Via: SIP/2.0/UDP192.168.5.44:5060;branch=z9hG4bK722c5489;rport
Max-Forwards: 70
From: "500"<sip:[email protected]>;tag=as08bd0ae7
To: <sip:[email protected]:5064>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: MyPBX
Date: Mon, 08 Oct 2012 09:20:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 1318995263 1318995263 IN IP4 192.168.5.44
s=Asterisk PBX SVN--r284M
c=IN IP4 192.168.5.44
t=0 0
m=audio 11122 RTP/AVP 0 8 9 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called 505
<--- SIP read from UDP:192.168.4.135:5064--->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP192.168.5.44:5060;branch=z9hG4bK722c5489;rport=5060
From: "500"<sip:[email protected]>;tag=as08bd0ae7
To: <sip:[email protected]:5064>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Yealink SIP-T26P 6.43.0.3000:15:65:12:0a:13
Content-Length: 0
MyPBX*CLI>
<--- SIP read fromUDP:192.168.4.135:5064 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP192.168.5.44:5060;branch=z9hG4bK722c5489;rport=5060
From: "500" <sip:[email protected]>;tag=as08bd0ae7
To:<sip:[email protected]:5064>;tag=1711960062
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5064>
User-Agent: Yealink SIP-T26P 6.43.0.3000:15:65:12:0a:13
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- SIP/505-00000015 is ringing
<--- Transmitting (no NAT) to192.168.4.18:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP192.168.4.18:5062;branch=z9hG4bK1993271819;received=192.168.4.18
From: "500"<sip:[email protected]>;tag=1071929271
To: <sip:[email protected]>;tag=as686c9fe2
Call-ID: [email protected]
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0
<--- SIP read fromUDP:192.168.4.135:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP192.168.5.44:5060;branch=z9hG4bK722c5489;rport=5060
From: "500"<sip:[email protected]>;tag=as08bd0ae7
To:<sip:[email protected]:5064>;tag=1711960062
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected]:5064>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE,CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T26P 6.43.0.3000:15:65:12:0a:13
Content-Length: 213
v=0
o=- 20094 20094 IN IP4 192.168.4.135
s=SDP data
c=IN IP4 192.168.4.135
t=0 0
m=audio 11786 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=ptime:20
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID0
Found audio description formattelephone-event for ID 101
[2012-10-08 01:21:05] NOTICE[549]:chan_sip.c:8805 process_sdp: p(17279900)->owner(8253986) Capabilities:
[2012-10-08 01:21:05] NOTICE[549]:chan_sip.c:8806 process_sdp: us - 0x110c (ulaw|alaw|g729|g722)
[2012-10-08 01:21:05] NOTICE[549]:chan_sip.c:8807 process_sdp: peer -0x4 (ulaw)
[2012-10-08 01:21:05] NOTICE[549]:chan_sip.c:8808 process_sdp: combined - 0x4 (ulaw)
[2012-10-08 01:21:05] NOTICE[549]:chan_sip.c:8810 process_sdp: owner:SIP/505-00000015 nativeformats:4readformat:4
writeformat:4
Non-codec capabilities (dtmf): us - 0x1(telephone-event), peer - 0x1 (telephone-event), combined - 0x1(telephone-event)
Peer audio RTP is at port192.168.4.135:11786
[2012-10-08 01:21:05] NOTICE[549]:chan_sip.c:8955 process_sdp: p(87769132)->owner:SIP/505-00000015nativeformats:4 readformat:4
writeformat:4
list_route: hop:<sip:[email protected]:5064>
set_destination: Parsing<sip:[email protected]:5064> for address/port to send to
set_destination: set destination to192.168.4.135, port 5064
Transmitting (no NAT) to192.168.4.135:5064:
ACK sip:[email protected]:5064 SIP/2.0
Via: SIP/2.0/UDP192.168.5.44:5060;branch=z9hG4bK30190331;rport
Max-Forwards: 70
From: "500" <sip:[email protected]>;tag=as08bd0ae7
To:<sip:[email protected]:5064>;tag=1711960062
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: MyPBX
Content-Length: 0