ASTERISK-21300: Asterisk is sending wrong codec order in the leg B of the call

[Home]

Summary: ASTERISK-21300: Asterisk is sending wrong codec order in the leg B of the call

Reporter: Jesus Tovar (jes_tovar) Labels:

Date Opened: 2013-03-19 02:19:20 Date Closed: 2013-04-09 15:31:00

Priority: Critical Regression?

Status: Closed/Complete Components: Channels/chan_sip/CodecHandling

Versions: 10.12.0 11.2.0 Frequency of
Occurrence

Related
Issues:

Environment: Centos 6.3 Linux rdns.vpdoc.com 2.6.32-279.22.1.el6.i686 #1 SMP Wed Feb 6 00:31:03 UTC 2013 i686 i686 i386 GNU/Linux Attachments: ( 0) SIP_Debug_PBX__V11_2_ticket.txt

Description: Basically the issue is focused on the codec order of the second leg of the call (PBX --> callee leg), where the PBX sets ulaw as first choice of the SDP whatever is the order in the configuration files of the peers, this order of codecs in the SDP makes the callee phone to pick up ulaw since it is already among the preference list of codecs. This issue is generating a permanent transcoding inside the server.

Whatever the codec order is in the Leg A of the call, the codec order in the Leg B is ulaw.

SDP_Caller:
Media Format: ITU-T G.711 PCMA
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.729

SDP_Callee:
Media Format: ITU-T G.711 PCMU
Media Format: ITU-T G.729
Media Format: ITU-T G.711 PCMA

Caller (g729/alaw/ulaw) ---> PBX --> (ulaw/alaw/g729) transcoding.
Caller (alaw/ulaw/g729) ---> PBX --> (ulaw/alaw/g729) transcoding.

Below there is a SIP Debug that shows clearly the failure:

Trace Information:
Number A (Caller): 3135
IP Address Caller: 10.1.1.237

Number B (Callee): 3088
IP Address Callee: 10.1.1.240

Sip.conf

disallow=all ; First disallow all codecs
allow=h264
allow=g729 ; Allow codecs in order of preference
allow=alaw
allow=ulaw

[3135]
type=peer
secret=7wq
canreinvite=no
host=dynamic
context=test
callerid="PBX CLIENT t6" <3135>
videosupport=yes
mailbox=3135@default
qualify=10000

[3088]
type=peer
secret=HIw
canreinvite=no
host=dynamic
context=test
callerid="PBX CLIENT t6" <3088>
videosupport=yes
mailbox=3088@default
qualify=10000

testing*CLI>
testing*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:10.1.1.240:32200 --->

<------------->

<--- SIP read from UDP:10.1.1.237:42936 --->
INVITE sip:3088@10.1.1.236:5066;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>
Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone>
Supported: replaces, timer, path
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5661 INVITE
User-Agent: Grandstream GXV3000 1.2.3.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 447

v=0
o=3135 8000 8000 IN IP4 10.1.1.237
s=SIP Call
c=IN IP4 10.1.1.237
t=0 0
m=audio 61432 RTP/AVP 8 0 18 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:40
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 61434 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
a=framerate:12
<------------->
--- (13 headers 18 lines) ---
Sending to 10.1.1.237:42936 (no NAT)
Using INVITE request as basis request - 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
Found peer '3135' for '3135' from 10.1.1.237:42936

<--- Reliably Transmitting (no NAT) to 10.1.1.237:42936 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05;received=10.1.1.237
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as1fa61e2e
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5661 INVITE
Server: PBX PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="PBX", nonce="4b547a92"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '87b5f796d6d01167924739f5bcb65c66@10.1.1.237' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.1.1.237:42936 --->
ACK sip:3088@10.1.1.236:5066;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bK8156bf5778454b05
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as1fa61e2e
Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone>
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5661 ACK
User-Agent: Grandstream GXV3000 1.2.3.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:10.1.1.237:42936 --->
INVITE sip:3088@10.1.1.236:5066;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>
Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone>
Supported: replaces, timer, path
Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:3088@10.1.1.236:5066;user=phone", nonce="4b547a92", response="199a546451f4d26bce7f8b623be1a32d"
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5662 INVITE
User-Agent: Grandstream GXV3000 1.2.3.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Content-Length: 447

v=0
o=3135 8000 8001 IN IP4 10.1.1.237
s=SIP Call
c=IN IP4 10.1.1.237
t=0 0
m=audio 61432 RTP/AVP 8 0 18 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=ptime:40
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 61434 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
a=framerate:12
<------------->
--- (14 headers 18 lines) ---
Sending to 10.1.1.237:42936 (no NAT)
Using INVITE request as basis request - 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
Found peer '3135' for '3135' from 10.1.1.237:42936
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|g729|h264), peer - audio=(ulaw|alaw|g729)/video=(h264)/text=(nothing), combined - (ulaw|alaw|g729|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.1.1.237:61432
Peer video RTP is at port 10.1.1.237:61434
Looking for 3088 in test (domain 10.1.1.236)
list_route: hop: <sip:3135@10.1.1.237:42936;transport=udp;user=phone>

<--- Transmitting (no NAT) to 10.1.1.237:42936 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5662 INVITE
Server: PBX PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3088@10.1.1.236:5066>
Content-Length: 0

<------------>
Audio is at 14418
Video is at 10.1.1.236:14292
Adding codec 100003 (ulaw) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.1.240:32200:
INVITE sip:3088@10.1.1.240:32200;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
Max-Forwards: 70
From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>
Contact: <sip:3135@10.1.1.236:5066>
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 102 INVITE
User-Agent: PBX PBX 11.2.1
Date: Mon, 18 Mar 2013 06:05:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 1112394555 1112394555 IN IP4 10.1.1.236
s=PBX PBX 11.2.1
c=IN IP4 10.1.1.236
b=CT:384
t=0 0
m=audio 14418 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 14292 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014
a=sendrecv

---

<--- SIP read from UDP:10.1.1.240:32200 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 102 INVITE
User-Agent: Grandstream GXV3000 1.2.3.7
Content-Length: 0

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

<--- SIP read from UDP:10.1.1.240:32200 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 102 INVITE
User-Agent: Grandstream GXV3000 1.2.3.7
Contact: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>

<--- Transmitting (no NAT) to 10.1.1.237:42936 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5662 INVITE
Server: PBX PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3088@10.1.1.236:5066>
Content-Length: 0

<------------>
Really destroying SIP dialog 'dc32e644caf312f6ab938814b3914792@10.1.1.240' Method: REGISTER

<--- SIP read from UDP:10.1.1.240:32200 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK19b27029
From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 102 INVITE
User-Agent: Grandstream GXV3000 1.2.3.7
Contact: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Type: application/sdp
Supported: replaces, timer, 100rel, path
Content-Length: 395

v=0
o=3088 8000 8000 IN IP4 10.1.1.240
s=SIP Call
c=IN IP4 10.1.1.240
t=0 0
m=audio 3084 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 3086 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0; sprop-parameter-sets=Z0KADJWgUH5A,aM4Ecg==
a=framerate:12
<------------->
--- (12 headers 16 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - (ulaw|alaw|g729|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.1.1.240:3084
Peer video RTP is at port 10.1.1.240:3086
list_route: hop: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>
set_destination: Parsing <sip:3088@10.1.1.240:32200;transport=udp;user=phone> for address/port to send to
set_destination: set destination to 10.1.1.240:32200
Transmitting (no NAT) to 10.1.1.240:32200:
ACK sip:3088@10.1.1.240:32200;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK45dec5cb
Max-Forwards: 70
From: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
To: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
Contact: <sip:3135@10.1.1.236:5066>
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 102 ACK
User-Agent: PBX PBX 11.2.1
Content-Length: 0

---
Audio is at 13140
Video is at 10.1.1.236:16842
Adding video codec 200004 (h264) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 10.1.1.237:42936 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKec101f6561e07d30;received=10.1.1.237
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5662 INVITE
Server: PBX PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:3088@10.1.1.236:5066>
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 1232695348 1232695348 IN IP4 10.1.1.236
s=PBX PBX 11.2.1
c=IN IP4 10.1.1.236
b=CT:384
t=0 0
m=audio 13140 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=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 16842 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014
a=sendrecv

<------------>

<--- SIP read from UDP:10.1.1.237:42936 --->
ACK sip:3088@10.1.1.236:5066 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.237:42936;branch=z9hG4bKcb05b5a399021ff0
From: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
To: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd
Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone>
Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:3088@10.1.1.236:5066;user=phone", nonce="4b547a92", response="199a546451f4d26bce7f8b623be1a32d"
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 5662 ACK
User-Agent: Grandstream GXV3000 1.2.3.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog 'bf1e2303-f567eb7e-f1b433c9@10.80.7.217' Method: REGISTER

<--- SIP read from UDP:10.1.1.240:32200 --->
BYE sip:3135@10.1.1.236:5066 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.240:32200;branch=z9hG4bKa26238e6e0617643
From: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
To: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 2839 BYE
User-Agent: Grandstream GXV3000 1.2.3.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Reason: SIP ;text="Onhook event"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 10.1.1.240:32200 (no NAT)
Scheduling destruction of SIP dialog '20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.1.1.240:32200 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.240:32200;branch=z9hG4bKa26238e6e0617643;received=10.1.1.240
From: <sip:3088@10.1.1.240:32200;transport=udp;user=phone>;tag=d137c401125656a4
To: "PBX CLIENT t6" <sip:3135@10.1.1.236:5066>;tag=as54d904ac
Call-ID: 20f762cf712c4a85090df4402cc71d56@10.1.1.236:5066
CSeq: 2839 BYE
Server: PBX PBX 11.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '87b5f796d6d01167924739f5bcb65c66@10.1.1.237' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:3135@10.1.1.237:42936;transport=udp;user=phone> for address/port to send to
set_destination: set destination to 10.1.1.237:42936
Reliably Transmitting (no NAT) to 10.1.1.237:42936:
BYE sip:3135@10.1.1.237:42936;transport=udp;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK121773f3
Max-Forwards: 70
From: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd
To: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 102 BYE
User-Agent: PBX PBX 11.2.1
Proxy-Authorization: Digest username="3135", realm="PBX", algorithm=MD5, uri="sip:10.1.1.236", nonce="4b547a92", response="d885ef09ee905ad79fb5c45e490084f1"
X-PBX-HangupCause: Normal Clearing
X-PBX-HangupCauseCode: 16
Content-Length: 0

---

<--- SIP read from UDP:10.1.1.237:42936 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.236:5066;branch=z9hG4bK121773f3
From: <sip:3088@10.1.1.236:5066;user=phone>;tag=as4b2e34fd
To: "VirtualHospital-RDT6" <sip:3135@10.1.1.236:5066;user=phone>;tag=2370b982d335c7c4
Call-ID: 87b5f796d6d01167924739f5bcb65c66@10.1.1.237
CSeq: 102 BYE
User-Agent: Grandstream GXV3000 1.2.3.7
Contact: <sip:3135@10.1.1.237:42936;transport=udp;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer, 100rel, path
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '87b5f796d6d01167924739f5bcb65c66@10.1.1.237' Method: ACK
testing*CLI> sip set debug off
SIP Debugging Disabled

Comments: By: Jesus Tovar (jes_tovar) 2013-03-19 02:20:39.255-0500

Tshark and Sip Debug information
By: Jesus Tovar (jes_tovar) 2013-03-19 02:23:26.224-0500

It is worth commenting this issue doesn't happened in version 1.8.X. As long as I migrated my servers to be in version 10.20 and up the problem with the wrong codec order is affecting my platform.
By: Rusty Newton (rnewton) 2013-03-22 14:46:10.575-0500

Please read through https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

Please remove all debug and file configuration from the description of the issue and re-attach as *separate* files.

It also looks like you included a wireshark capture in the file you did attach (which is also redundant with some of the information in the description?)

Please separate out the wireshark pcap into a .pcap file and attach it to the issue.

By: Rusty Newton (rnewton) 2013-04-09 15:30:46.560-0500

Reporter hasn't cleaned up the issue to follow the guidelines or provided any additional debug.

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines

To look into this further we would need a proper pcap attached, plus an Asterisk log (from 1.8 or 11) that includes SIP debug showing the issue along with VERBOSE and DEBUG messages enabled and turned up to level 5.