Просьба ногами не пинать, это мой первый опыт с Астериском...

Клиент к серверу цепляется, авторизация проходит, аудио работает, видео - нет. Из видеокодеков на клиенте оставил только H.263+ (оди плюс).

На сервере:
helios*CLI> core show codecs video
Disclaimer: this command is for informational purposes only.
It does not indicate anything about your configuration.
INT BINARY HEX TYPE NAME DESC
--------------------------------------------------------------------------------
262144 (1 << 18) (0x40000) video h261 (H.261 Video)
524288 (1 << 19) (0x80000) video h263 (H.263 Video)
1048576 (1 << 20) (0x100000) video h263p (H.263+ Video)
2097152 (1 << 21) (0x200000) video h264 (H.264 Video)

Настройки SIP:

helios*CLI> sip show settings
Global Settings:
----------------
SIP Port: 5060
Bindaddress: 192.168.1.251
Videosupport: Yes
AutoCreatePeer: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: No
Promsic. redir: No
SIP domain support: Yes
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm ves_lan
Realm. auth: No
Always auth rejects: No
Call limit peers only: No
Direct RTP setup: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
T38 fax pt UDPTL: No
RFC2833 Compensation: No
SIP realtime: Disabled

Global Signalling Settings:
---------------------------
Codecs: 0x3c000e (gsm|ulaw|alaw|h261|h263|h263p|h264)
Codec Order: ()
T1 minimum: 100
Relax DTMF: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No

Default Settings:
-----------------
Context: Office
Nat: No
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk

----

Разрешенные кодеки:
allow=h261
allow=h263
allow=h263p
allow=h264

Конфиг пиров:

[300]; video1
type=friend
username=300
secret=секретное_слово
host=dynamic
context=office
callerid=Admin <300>
disallow=all ; better for custom-tunning codec selection
canreinvite=no
allow=ulaw
allow=alaw
allow=gsm
allow=h261
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
allow=h264
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833
canreinvite=no ; canreinvite must be set to 'no'

Всего 3 пира с видео, параметры у них одинаковые.

sip set debug

интересует примерно такой кусок

 Capabilities: us - 0x20000c (ulaw|alaw|h264), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)

после disallow должно сразу идти allow, что там между ними canreinvite делает у вас?

<------------->
--- (13 headers 16 lines) ---
Sending to 192.168.1.100 : 54158 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found RTP video format 115
Peer audio RTP is at port 192.168.1.100:61340
Found audio description format telephone-event for ID 101
Found video description format H263-1998 for ID 115
Capabilities: us - 0x3c000e (gsm|ulaw|alaw|h261|h263|h263p|h264), peer - audio=0 x10000e (gsm|ulaw|alaw|h263p)/video=0x100000 (h263p), combined - (ulaw|alaw|gsm| h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone -event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.100:61340
Peer video RTP is at port 192.168.1.100:8498

в sip.conf в [general] добавь videosupport=yes

theoc,

helios*CLI> sip show settings
Global Settings:
----------------
SIP Port: 5060
Bindaddress: 192.168.1.251
Videosupport: Yes
AutoCreatePeer: No
Allow unknown access: No
.......

После того, как в sip.conf сделал

[GENERAL]
........
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=ilbc ; see doc/rtp-packetization for framing options
allow=h261
allow=h263
allow=h263p
allow=h264
........
[300]
type=friend
username=300
secret=секретное_слово
host=dynamic
context=office
canreinvite=no
callerid=Admin <300>
disallow=all ; better for custom-tunning codec selection
allow=ulaw
allow=alaw
allow=gsm
allow=h261
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
allow=h264
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833
canreinvite=no ; canreinvite must be set to 'no'

А в клиентах оставил только H263+ (1998), при попытке инициировать видео в лог стало валиться:
[Jun 3 15:03:31] NOTICE[31880] rtp.c: Unknown RTP codec 126 received from '192.168.1.100'
А на клиенте надпись - куд нот старт видео

Что за кодек такой 126?

мои идеи закончены. остаются танцы с бубном. можно попробовать в пире запретить все видеокодеки кроме h263p

После того, как у клиентов оставил только H.263+ (1998) из видео и GSM из аудио, при нажатии кнопки Start видео в клиенте, клиент говорит "Куд нот старт видео", а в дебаге сипа:

<--- SIP read from 192.168.1.100:8738 --->
INVITE sip:301@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:8738;branch=z9hG4bK-d87543-504fc6335274a151-1--d8 7543-;rport
Max-Forwards: 70
Contact: <sip:300@192.168.1.100:8738>
To: "test1"<sip:301@192.168.1.251>;tag=as106d9be6
From: "Admin"<sip:300@192.168.1.251>;tag=404a547a
Call-ID: NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.
CSeq: 9 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O
Content-Type: application/sdp
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="26337d22",uri= "sip:301@192.168.1.251",response="70d5ddabdcb4fbfa9d397a250665e000",algorithm=MD 5
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 441

v=0
o=- 0 9 IN IP4 192.168.1.100
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.100
t=0 0
m=audio 27554 RTP/AVP 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:8711D47F2DAE47DBAFFCCF14E3FFA85A
m=video 4530 RTP/AVP 115
a=alt:1 1 : Lcr+J9it 5rH2G/pE 192.168.1.100 4530
a=fmtp:115 QCIF=2 MAXBR=10485
a=rtpmap:115 H263-1998/90000
a=sendrecv
a=x-rtp-session-id:F188CE49CB0C4BF09755DDDA3F70D467

<------------->
--- (13 headers 16 lines) ---
Sending to 192.168.1.100 : 8738 (NAT)
Found RTP audio format 3
Found RTP audio format 101
Found RTP video format 115
Peer audio RTP is at port 192.168.1.100:27554
Found audio description format telephone-event for ID 101
Found video description format H263-1998 for ID 115
Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - audio=0x100002 (gsm|h26 3p)/video=0x100000 (h263p), combined - (gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone -event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.100:27554
Peer video RTP is at port 192.168.1.100:4530

<--- Transmitting (NAT) to 192.168.1.100:8738 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.100:8738;branch=z9hG4bK-d87543-504fc6335274a151-1--d8 7543-;rport;received=192.168.1.100
From: "Admin"<sip:300@192.168.1.251>;tag=404a547a
To: "test1"<sip:301@192.168.1.251>;tag=as106d9be6
Call-ID: NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.
CSeq: 9 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.251>
Content-Length: 0


<------------>
Audio is at 192.168.1.251 port 18454
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.1.100:8738 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:8738;branch=z9hG4bK-d87543-504fc6335274a151-1--d8 7543-;rport;received=192.168.1.100
From: "Admin"<sip:300@192.168.1.251>;tag=404a547a
To: "test1"<sip:301@192.168.1.251>;tag=as106d9be6
Call-ID: NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.
CSeq: 9 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.251>
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 31880 31887 IN IP4 192.168.1.251
s=session
c=IN IP4 192.168.1.251
t=0 0
m=audio 18454 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
helios*CLI>
<--- SIP read from 192.168.1.100:8738 --->
ACK sip:301@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:8738;branch=z9hG4bK-d87543-ed7f77104f4d9c11-1--d8 7543-;rport
Max-Forwards: 70
Contact: <sip:300@192.168.1.100:8738>
To: "test1"<sip:301@192.168.1.251>;tag=as106d9be6
From: "Admin"<sip:300@192.168.1.251>;tag=404a547a
Call-ID: NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.
CSeq: 9 ACK
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="26337d22",uri= "sip:301@192.168.1.251",response="70d5ddabdcb4fbfa9d397a250665e000",algorithm=MD 5
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
helios*CLI>
<--- SIP read from 192.168.1.100:8738 --->
BYE sip:301@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.100:8738;branch=z9hG4bK-d87543-c870b82467536233-1--d8 7543-;rport
Max-Forwards: 70
Contact: <sip:300@192.168.1.100:8738>
To: "test1"<sip:301@192.168.1.251>;tag=as106d9be6
From: "Admin"<sip:300@192.168.1.251>;tag=404a547a
Call-ID: NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.
CSeq: 10 BYE
Proxy-Authorization: Digest username="300",realm="ves_lan",nonce="26337d22",uri= "sip:301@192.168.1.251",response="0c8cc80ba17b3539f129f4d1eec727ff",algorithm=MD 5
User-Agent: eyeBeam release 1004p stamp 31962
Reason: SIP;description="User Hung Up"
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.1.100 : 8738 (NAT)

<--- Transmitting (NAT) to 192.168.1.100:8738 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.100:8738;branch=z9hG4bK-d87543-c870b82467536233-1--d8 7543-;rport;received=192.168.1.100
From: "Admin"<sip:300@192.168.1.251>;tag=404a547a
To: "test1"<sip:301@192.168.1.251>;tag=as106d9be6
Call-ID: NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.
CSeq: 10 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:301@192.168.1.251>
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2131f03c62d35d9a7879986911e53a21@192.168.1 .251' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:301@192.168.1.13:36796;rinstance=5a1851d8b0233bd7> for address/port to send to
set_destination: set destination to 192.168.1.13, port 36796
Reliably Transmitting (NAT) to 192.168.1.13:36796:
BYE sip:301@192.168.1.13:36796;rinstance=5a1851d8b0233bd7 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK78632c27
From: "Admin"<sip:300@192.168.1.251>;tag=as4ce531a3
To: <sip:301@192.168.1.13:36796;rinstance=5a1851d8b0233bd7>;tag=1c1b1a1e
Call-ID: 2131f03c62d35d9a7879986911e53a21@192.168.1.251
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
helios*CLI>
<--- SIP read from 192.168.1.13:36796 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK78632c27
Contact: <sip:301@192.168.1.13:36796;rinstance=5a1851d8b0233bd7>
To: <sip:301@192.168.1.13:36796;rinstance=5a1851d8b0233bd7>;tag=1c1b1a1e
From: "Admin"<sip:300@192.168.1.251>;tag=as4ce531a3
Call-ID: 2131f03c62d35d9a7879986911e53a21@192.168.1.251
CSeq: 103 BYE
User-Agent: eyeBeam release 1004p stamp 31962
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2131f03c62d35d9a7879986911e53a21@192.168.1.251' Me thod: ACK
Really destroying SIP dialog 'NTBmNzEyYWFhNTE3NWY1OWVjZmVjZGEyYTJjY2EzNTA.' Meth od: BYE

В логе на тему кодеков чисто.

ну как это чисто? грязно ведь!

Capabilities: us - 0x20000e (gsm|ulaw|alaw|h264), peer - audio=0x100002 (gsm|h26 3p)/video=0x100000 (h263p), combined - (gsm)
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