正常的codec协商包
<--- SIP read from UDP:192.168.4.18:5067--->
INVITE sip:301@192.168.4.122 SIP/2.0
Via: SIP/2.0/UDP192.168.4.18:5067;branch=z9hG4bK424774178
From: "300"<sip:300@192.168.4.122>;tag=778739804
To: <sip:301@192.168.4.122>
Call-ID: 616179575@192.168.4.18
CSeq: 2 INVITE
Contact: <sip:300@192.168.4.18:5067>
Authorization: Digestusername="300", realm="asterisk",nonce="61bc3751", uri="sip:301@192.168.4.122",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 -616179575@192.168.4.18
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:300@192.168.4.18:5067>
<--- 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:300@192.168.4.122>;tag=778739804
To: <sip:301@192.168.4.122>
Call-ID: 616179575@192.168.4.18
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:301@192.168.4.122>
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:301@192.168.4.122>;tag=as5a4d8aee
To: "300"<sip:300@192.168.4.122>;tag=43927923
Call-ID: 15125666@192.168.4.18
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:301@192.168.4.26:5066 SIP/2.0
Via: SIP/2.0/UDP192.168.4.122:5060;branch=z9hG4bK7ca8a118;rport
Max-Forwards: 70
From: "300"<sip:300@192.168.4.122>;tag=as09e7c96d
To: <sip:301@192.168.4.26:5066>
Contact: <sip:300@192.168.4.122>
Call-ID: 0dc2a4c77788fcb667ae0f3149dbbb45@192.168.4.122
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:300@192.168.4.122>;tag=as20c8c805
To: <sip:301@192.168.4.26:5066>
Call-ID: 4a3e263c19a7376b16df6eae68959d3e@192.168.4.122
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:300@192.168.4.122>;tag=as20c8c805
To:<sip:301@192.168.4.26:5066>;