[Asterisk-video] Unsupported SDP media type

Tue Oct 11 02:29:52 CDT 2011

Hi ,

I am attaching the sip debug messages enabled on asterisk server .the
present status is that when i call from Mirial softphone ,the call gets
established and gets disconnected after a while .i can't see video on mcuWeb
page .i think i have done all things correctly but i am stuck with this
thing .please help me out of this problem .

INVITE sip:300 at 192.168.115.29:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=
z9hG4bK504e1a07
Max-Forwards: 70
From: "205" <sip:205 at 192.168.115.53>;tag=as03ab7ca9
To: <sip:300 at 192.168.115.29:5060>
Contact: <sip:205 at 192.168.115.53:5060>
Call-ID: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.6.0
Date: Tue, 11 Oct 2011 06:54:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 363

v=0
o=root 1739466419 1739466419 IN IP4 192.168.115.53
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.115.53
b=CT:384
t=0 0
m=audio 14856 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 15692 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

---

<--- SIP read from UDP:192.168.115.29:5060 --->
SIP/2.0 100 Trying
Content-Length: 0
To: <sip:300 at 192.168.115.29:5060>
Cseq: 102 INVITE
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07
Server: Glassfish_SIP_1.0.0
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
From: "205" <sip:205 at 192.168.115.53>;tag=as03ab7ca9

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

<--- SIP read from UDP:192.168.115.29:5060 --->
SIP/2.0 180 Ringing
Content-Length: 0
To: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
Contact: <sip:192.168.115.29:5070;fid=server_1>
Cseq: 102 INVITE
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07
From: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
Server: Glassfish_SIP_1.0.0

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

<--- Transmitting (no NAT) to 192.168.115.40:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.115.40:5060
;branch=z9hG4bK-d7358366c5-DL;received=192.168.115.40
From: "205" <sip:205 at 192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8
To: <sip:300 at 192.168.115.53;user=phone>;tag=as6ba67fa9
Call-ID: DLd873cc4431-1078986205 at fstl-312
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:300 at 192.168.115.53:5060>
Content-Length: 0

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

<--- SIP read from UDP:192.168.115.29:5060 --->
SIP/2.0 200 Ok
Content-Length: 245
To: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
Contact: <sip:192.168.115.29:5070;fid=server_1>
Cseq: 102 INVITE
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07
Content-Type: application/sdp
From: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
Server: Glassfish_SIP_1.0.0

v=0
o=- 0 0 IN IP4 192.168.115.29
s=MediaMixerSession
c=IN IP4 192.168.115.29
t=0 0
m=audio 60014 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 35814 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428021
<------------->
--- (10 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found RTP video format 99
Found video description format H264 for ID 99
Capabilities: us - 0x30000c (ulaw|alaw|h263p|h264), peer - audio=0xc
(ulaw|alaw)/video=0x200000 (h264)/text=0x0 (nothing), combined - 0x20000c
(ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.115.29:60014
Peer video RTP is at port 192.168.115.29:35814
list_route: hop: <sip:192.168.115.29:5070;fid=server_1>
set_destination: Parsing <sip:192.168.115.29:5070;fid=server_1> for
address/port to send to
set_destination: set destination to 192.168.115.29:5070
Transmitting (no NAT) to 192.168.115.29:5070:
ACK sip:192.168.115.29:5070;fid=server_1 SIP/2.0
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK07561f41
Max-Forwards: 70
From: "205" <sip:205 at 192.168.115.53>;tag=as03ab7ca9
To: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
Contact: <sip:205 at 192.168.115.53:5060>
Call-ID: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0

---
Audio is at 5060
Video is at 192.168.115.53:5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.115.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.115.40:5060
;branch=z9hG4bK-d7358366c5-DL;received=192.168.115.40
From: "205" <sip:205 at 192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8
To: <sip:300 at 192.168.115.53;user=phone>;tag=as6ba67fa9
Call-ID: DLd873cc4431-1078986205 at fstl-312
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:300 at 192.168.115.53:5060>
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 797835425 797835425 IN IP4 192.168.115.53
s=Asterisk PBX 1.8.6.0
c=IN IP4 192.168.115.53
b=CT:384
t=0 0
m=audio 10346 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 12196 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

<------------>
    -- Locally bridging SIP/205-00000002 and SIP/mcuWeb-00000003

<--- SIP read from UDP:192.168.115.40:5060 --->
ACK sip:300 at 192.168.115.53:5060 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.115.40:5060;branch=z9hG4bK-818b523671-DL
To: <sip:300 at 192.168.115.53;user=phone>;tag=as6ba67fa9
From: "205" <sip:205 at 192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8
Call-ID: DLd873cc4431-1078986205 at fstl-312
Max-Forwards: 70
Contact: "205" <sip:205 at 192.168.115.40:5060>
Content-Length: 0

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

<--- SIP read from UDP:192.168.115.29:5060 --->
SIP/2.0 200 Ok
Content-Length: 245
To: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
Contact: <sip:192.168.115.29:5070;fid=server_1>
Cseq: 102 INVITE
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK504e1a07
Content-Type: application/sdp
From: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
Server: Glassfish_SIP_1.0.0

v=0
o=- 0 0 IN IP4 192.168.115.29
s=MediaMixerSession
c=IN IP4 192.168.115.29
t=0 0
m=audio 60014 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
m=video 35814 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428021
<------------->
--- (10 headers 11 lines) ---
set_destination: Parsing <sip:192.168.115.29:5070;fid=server_1> for
address/port to send to
set_destination: set destination to 192.168.115.29:5070
Transmitting (no NAT) to 192.168.115.29:5070:
ACK sip:192.168.115.29:5070;fid=server_1 SIP/2.0
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK4559c232
Max-Forwards: 70
From: "205" <sip:205 at 192.168.115.53>;tag=as03ab7ca9
To: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
Contact: <sip:205 at 192.168.115.53:5060>
Call-ID: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.6.0
Content-Length: 0

<--- SIP read from UDP:192.168.115.29:5060 --->
BYE sip:205 at 192.168.115.53:5060 SIP/2.0
Max-Forwards: 69
Content-Length: 0
To: "205"<sip:205 at 192.168.115.53>;
tag=as03ab7ca9
Contact: <sip:192.168.115.29:5070;fid=server_1>
Cseq: 2 BYE
Via: SIP/2.0/UDP 192.168.115.29:5070
;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01
From: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.115.29:5070 (no NAT)
Scheduling destruction of SIP dialog '
687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060' in 32000 ms (Method:
BYE)

<--- Transmitting (no NAT) to 192.168.115.29:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.115.29:5070
;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01;received=192.168.115.29
From: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
To: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Call-ID: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
CSeq: 2 BYE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'DLd873cc4431-1078986205 at fstl-312' in
32000 ms (Method: OPTIONS)
set_destination: Parsing <sip:205 at 192.168.115.40:5060> for address/port to
send to
set_destination: set destination to 192.168.115.40:5060
Reliably Transmitting (no NAT) to 192.168.115.40:5060:
BYE sip:205 at 192.168.115.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.115.53:5060;branch=z9hG4bK22de4c9c
Max-Forwards: 70
From: <sip:300 at 192.168.115.53;user=phone>;tag=as6ba67fa9
To: "205" <sip:205 at 192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8
Call-ID: DLd873cc4431-1078986205 at fstl-312
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.6.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

<--- SIP read from UDP:192.168.115.40:5060 --->
SIP/2.0 200 OK
From: <sip:300 at 192.168.115.53;user=phone>;tag=as6ba67fa9;epid=021776A8
Call-ID: DLd873cc4431-1078986205 at fstl-312
CSeq: 102 BYE
Via: SIP/2.0/UDP 192.168.115.53:5060
;branch=z9hG4bK22de4c9c;received=192.168.115.53
To: "205" <sip:205 at 192.168.115.53>;tag=DLb62dc55e0d;epid=021776A8
Allow:
INVITE,CANCEL,ACK,OPTIONS,INFO,SUBSCRIBE,NOTIFY,BYE,MESSAGE,UPDATE,REFER
Contact: "205" <sip:205 at 192.168.115.40:5060>
User-Agent: Dylogic Mirial 7.0.54
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 'DLd873cc4431-1078986205 at fstl-312' Method:
OPTIONS

<--- SIP read from UDP:192.168.115.29:5060 --->
BYE sip:205 at 192.168.115.53:5060 SIP/2.0
Max-Forwards: 69
Content-Length: 0
Contact: <sip:192.168.115.29:5070;fid=server_1>
To: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Cseq: 2 BYE
Via: SIP/2.0/UDP 192.168.115.29:5070
;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
From: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.115.29:5070 (no NAT)

<--- Transmitting (no NAT) to 192.168.115.29:5070 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.115.29:5070
;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01;received=192.168.115.29
From: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
To: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Call-ID: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
CSeq: 2 BYE
Server: Asterisk PBX 1.8.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<--- SIP read from UDP:192.168.115.29:5060 --->
BYE sip:205 at 192.168.115.53:5060 SIP/2.0
Max-Forwards: 69
Content-Length: 0
Contact: <sip:192.168.115.29:5070;fid=server_1>
To: "205"<sip:205 at 192.168.115.53>;tag=as03ab7ca9
Cseq: 2 BYE
Via: SIP/2.0/UDP 192.168.115.29:5070
;branch=z9hG4bKdaaccb789521ce3e4a69b8b9f63068bbfa01
Call-Id: 687d923112d7df683bf0cb08115c1d7d at 192.168.115.53:5060
From: <sip:300 at 192.168.115.29:5060>;tag=gtmjcomk-b
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