asterisk sip codec协商

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>;