Проблемы с голосом

Question

Всем привет.

Asterisk установлена плата TDM808P. К ней подключен GSM модем Teleofice. Также к * подключено 2 SIP клиента (X-lite и самописный SIP клиент ). Звонок с GSM модема проброшен на SIP клиентов. В итоге: если звонить на X-lite из GSM сети, звук отличный в обе стороны. А при звонке на самописный клиент, то в сторону GSM качество речи плохое (бульканье), в сторону SIP клиента качество хорошее. Но, что самое интересное, запись разговоров, которая ведется на * получается хорошая в обоих случаях. Подскажите куда копать?

Проблемный звонок:

localhost*CLI> 
localhost*CLI> sip set debug peer  102
SIP Debugging Enabled for IP: 10.0.0.5
Really destroying SIP dialog 'RbViVEUp-1333971970875@Dispatcher1' Method: REGISTER
localhost*CLI> sip set debug peer  102
SIP Debugging Enabled for IP: 10.0.0.5
    -- Starting simple switch on 'DAHDI/7-1'
[Apr  9 15:46:56] ERROR[17731]: callerid.c:562 callerid_feed: No start bit found in fsk data.
[Apr  9 15:46:56] WARNING[17731]: chan_dahdi.c:1725 my_get_callerid: Failed to decode CallerID
    -- Executing [s@incoming-from-pstn:1] Wait("DAHDI/7-1", "1") in new stack
    -- Executing [s@incoming-from-pstn:2] Answer("DAHDI/7-1", "") in new stack
    -- Executing [s@incoming-from-pstn:3] NoOp("DAHDI/7-1", "CALLER_ID === "" <>") in new stack
    -- Executing [s@incoming-from-pstn:4] NoOp("DAHDI/7-1", "2012-04-09") in new stack
    -- Executing [s@incoming-from-pstn:5] Set("DAHDI/7-1", "RECORD_FILE_POSTFIX=") in new stack
    -- Executing [s@incoming-from-pstn:6] Set("DAHDI/7-1", "RECORDS_DIRNAME=/var/spool/asterisk/records/2012-04-09") in new stack
    -- Executing [s@incoming-from-pstn:7] System("DAHDI/7-1", "mkdir -p /var/spool/asterisk/records/2012-04-09") in new stack
    -- Executing [s@incoming-from-pstn:8] Wait("DAHDI/7-1", "1") in new stack
    -- Executing [s@incoming-from-pstn:9] Playback("DAHDI/7-1", "taxisounds/hello") in new stack
    -- <DAHDI/7-1> Playing 'taxisounds/hello.gsm' (language 'ru')

<--- SIP read from UDP:10.0.0.5:5060 --->


<------------->
    -- Executing [s@incoming-from-pstn:10] Set("DAHDI/7-1", "MONITOR_FILENAME=/var/spool/asterisk/records/2012-04-09/15-47-03_") in new stack
    -- Executing [s@incoming-from-pstn:11] Queue("DAHDI/7-1", "taxi-operators,tr,,,300") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.5:5060:
INVITE sip:102@10.0.0.5:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK28fed8f5
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.1>;tag=as3bf168b2
To: <sip:102@10.0.0.5:5060;transport=UDP>
Contact: <sip:asterisk@10.0.0.1:5060>
Call-ID: 6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.0
Date: Mon, 09 Apr 2012 11:47:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 275

v=0
o=root 1352973717 1352973717 IN IP4 10.0.0.1
s=Asterisk PBX 1.8.7.0
c=IN IP4 10.0.0.1
t=0 0
m=audio 19536 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/102

<--- SIP read from UDP:10.0.0.5:5060 --->
SIP/2.0 180 Ringing
From: "asterisk" <sip:asterisk@10.0.0.1>;tag=as3bf168b2
Call-ID: 6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK28fed8f5
To: <sip:102@10.0.0.5:5060;transport=UDP>;tag=XhFClljEzI
Contact: <sip:10.0.0.5:5060;transport=UDP>

<------------->
--- (7 headers 0 lines) ---
    -- SIP/102-00004380 is ringing
localhost*CLI> rtp set debug on
RTP Debugging Enabled

<--- SIP read from UDP:10.0.0.5:5060 --->
SIP/2.0 200 OK
From: "asterisk" <sip:asterisk@10.0.0.1>;tag=as3bf168b2
Call-ID: 6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK28fed8f5
To: <sip:102@10.0.0.5:5060;transport=UDP>;tag=XhFClljEzI
Content-Length: 202
Content-Type: application/sdp
Contact: <sip:10.0.0.5:5060;transport=UDP>

v=0
o=user1 136363567 527806460 IN IP4 10.0.0.5
s=-
c=IN IP4 10.0.0.5
t=0 0
m=audio 8000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
--- (9 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.5:8000
list_route: hop: <sip:10.0.0.5:5060;transport=UDP>
set_destination: Parsing <sip:10.0.0.5:5060;transport=UDP> for address/port to send to
set_destination: set destination to 10.0.0.5:5060
Transmitting (no NAT) to 10.0.0.5:5060:
ACK sip:10.0.0.5:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK4d31c358
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.1>;tag=as3bf168b2
To: <sip:102@10.0.0.5:5060;transport=UDP>;tag=XhFClljEzI
Contact: <sip:asterisk@10.0.0.1:5060>
Call-ID: 6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.0
Content-Length: 0


---
    -- SIP/102-00004380 answered DAHDI/7-1
  == Begin MixMonitor Recording DAHDI/7-1
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040012, ts 000160, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040013, ts 000320, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040014, ts 000480, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040015, ts 000640, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040016, ts 000800, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040017, ts 000960, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040018, ts 001120, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040019, ts 001280, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040020, ts 001440, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040021, ts 001600, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040022, ts 001760, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040023, ts 001920, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040024, ts 002080, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040025, ts 002240, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040026, ts 002400, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040027, ts 002560, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040028, ts 002720, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040029, ts 002880, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040030, ts 003040, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040031, ts 003200, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040032, ts 003360, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040033, ts 003520, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040034, ts 003680, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040035, ts 003840, len 000160)
Sent RTP packet to      10.0.0.5:8000 (type 00, seq 040036, ts 004000, len 000160)

Got  RTP packet from    10.0.0.5:8000 (type 00, seq 017791, ts 053280, len 000160)


localhost*CLI> rtp set debug off
Got  RTP packet from    10.0.0.5:8000 (type 00, seq 017833, ts 060000, len 000160)
RTP Debugging Disabled

<--- SIP read from UDP:10.0.0.5:5060 --->


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

<--- SIP read from UDP:10.0.0.5:5060 --->


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

<--- SIP read from UDP:10.0.0.5:5060 --->
BYE sip:asterisk@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:5060;rport;branch=z9hG4bKhnbBx5NFk
Max-Forwards: 70
To: <sip:asterisk@10.0.0.1>;tag=as3bf168b2
From: <sip:102@10.0.0.5:5060;transport=UDP>;tag=XhFClljEzI
Call-ID: 6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060
CSeq: 666986036 BYE
Authorization: Digest username="102", realm="asterisk", nonce="63809b3d", uri="sip:10.0.0.1", response="1e8e2e54dd6ebf9f2f6dd3629e621081"

<------------->
--- (8 headers 0 lines) ---
Sending to 10.0.0.5:5060 (no NAT)
Scheduling destruction of SIP dialog '6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 10.0.0.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.5:5060;branch=z9hG4bKhnbBx5NFk;received=10.0.0.5;rport=5060
From: <sip:102@10.0.0.5:5060;transport=UDP>;tag=XhFClljEzI
To: <sip:asterisk@10.0.0.1>;tag=as3bf168b2
Call-ID: 6dceec570cd04a0e61e5608e76d998f8@10.0.0.1:5060
CSeq: 666986036 BYE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
  == Spawn extension (incoming-from-pstn, s, 11) exited non-zero on 'DAHDI/7-1'
    -- Hanging up on 'DAHDI/7-1'
    -- Hungup 'DAHDI/7-1'
  == MixMonitor close filestream
  == End MixMonitor Recording DAHDI/7-1
localhost*CLI> exit



Хороший звонок:


localhost*CLI> 
    -- Starting simple switch on 'DAHDI/7-1'


<--- SIP read from UDP:10.0.0.5:8762 --->
SUBSCRIBE sip:102@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:8762;branch=z9hG4bK-d87543-be1a74450e0eff5a-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:102@10.0.0.5:8762>
To: <sip:102@10.0.0.1>
From: <sip:102@10.0.0.1>;tag=0133f46e
Call-ID: 4d12cb78eb3e1963NWUxYzUzZGQwYWM3NjdkY2UzMzJmMzY1ODFiNWM2MjQ.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1003l stamp 30942
Event: message-summary
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 10.0.0.5:8762 (no NAT)
list_route: hop: <sip:102@10.0.0.5:8762>
Found peer '102' for '102' from 10.0.0.5:8762

<--- Transmitting (no NAT) to 10.0.0.5:8762 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.5:8762;branch=z9hG4bK-d87543-be1a74450e0eff5a-1--d87543-;received=10.0.0.5;rport=8762
From: <sip:102@10.0.0.1>;tag=0133f46e
To: <sip:102@10.0.0.1>;tag=as0edff57a
Call-ID: 4d12cb78eb3e1963NWUxYzUzZGQwYWM3NjdkY2UzMzJmMzY1ODFiNWM2MjQ.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0cd47680"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '4d12cb78eb3e1963NWUxYzUzZGQwYWM3NjdkY2UzMzJmMzY1ODFiNWM2MjQ.' in 32000 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:10.0.0.5:8762 --->
SUBSCRIBE sip:102@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.5:8762;branch=z9hG4bK-d87543-b25b702d9a2f5755-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:102@10.0.0.5:8762>
To: <sip:102@10.0.0.1>
From: <sip:102@10.0.0.1>;tag=0133f46e
Call-ID: 4d12cb78eb3e1963NWUxYzUzZGQwYWM3NjdkY2UzMzJmMzY1ODFiNWM2MjQ.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1003l stamp 30942
Authorization: Digest username="102",realm="asterisk",nonce="0cd47680",uri="sip:102@10.0.0.1",response="a9731a580dc7c52bb79b803c4ba3c334",algorithm=MD5
Event: message-summary
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 10.0.0.5:8762 (no NAT)
Found peer '102' for '102' from 10.0.0.5:8762

<--- Transmitting (no NAT) to 10.0.0.5:8762 --->
SIP/2.0 404 Not found (no mailbox)
Via: SIP/2.0/UDP 10.0.0.5:8762;branch=z9hG4bK-d87543-b25b702d9a2f5755-1--d87543-;received=10.0.0.5;rport=8762
From: <sip:102@10.0.0.1>;tag=0133f46e
To: <sip:102@10.0.0.1>;tag=as0edff57a
Call-ID: 4d12cb78eb3e1963NWUxYzUzZGQwYWM3NjdkY2UzMzJmMzY1ODFiNWM2MjQ.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Apr  9 15:44:45] NOTICE[24722]: chan_sip.c:24173 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 102
Really destroying SIP dialog '4d12cb78eb3e1963NWUxYzUzZGQwYWM3NjdkY2UzMzJmMzY1ODFiNWM2MjQ.' Method: SUBSCRIBE
    -- Executing [s@incoming-from-pstn:2] Answer("DAHDI/7-1", "") in new stack
    -- Executing [s@incoming-from-pstn:3] NoOp("DAHDI/7-1", "CALLER_ID === "" <>") in new stack
    -- Executing [s@incoming-from-pstn:4] NoOp("DAHDI/7-1", "2012-04-09") in new stack
    -- Executing [s@incoming-from-pstn:5] Set("DAHDI/7-1", "RECORD_FILE_POSTFIX=") in new stack
    -- Executing [s@incoming-from-pstn:6] Set("DAHDI/7-1", "RECORDS_DIRNAME=/var/spool/asterisk/records/2012-04-09") in new stack
    -- Executing [s@incoming-from-pstn:7] System("DAHDI/7-1", "mkdir -p /var/spool/asterisk/records/2012-04-09") in new stack
    -- Executing [s@incoming-from-pstn:8] Wait("DAHDI/7-1", "1") in new stack
    -- Executing [s@incoming-from-pstn:9] Playback("DAHDI/7-1", "taxisounds/hello") in new stack
    -- <DAHDI/7-1> Playing 'taxisounds/hello.gsm' (language 'ru')
    -- Executing [s@incoming-from-pstn:10] Set("DAHDI/7-1", "MONITOR_FILENAME=/var/spool/asterisk/records/2012-04-09/15-44-50_") in new stack
    -- Executing [s@incoming-from-pstn:11] Queue("DAHDI/7-1", "taxi-operators,tr,,,300") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.5:8762:
INVITE sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1039cf44
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.1>;tag=as09f98f70
To: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>
Contact: <sip:asterisk@10.0.0.1:5060>
Call-ID: 777b7a812c3fa1907620794b55b1abcf@10.0.0.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.0
Date: Mon, 09 Apr 2012 11:44:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 273

v=0
o=root 324401610 324401610 IN IP4 10.0.0.1
s=Asterisk PBX 1.8.7.0
c=IN IP4 10.0.0.1
t=0 0
m=audio 11056 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/102

<--- SIP read from UDP:10.0.0.5:8762 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1039cf44
Contact: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>
To: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>;tag=da5c7871
From: "asterisk"<sip:asterisk@10.0.0.1>;tag=as09f98f70
Call-ID: 777b7a812c3fa1907620794b55b1abcf@10.0.0.1:5060
CSeq: 102 INVITE
User-Agent: X-Lite release 1003l stamp 30942
Content-Length: 0

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


<--- SIP read from UDP:10.0.0.5:8762 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK1039cf44
Contact: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>
To: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>;tag=da5c7871
From: "asterisk"<sip:asterisk@10.0.0.1>;tag=as09f98f70
Call-ID: 777b7a812c3fa1907620794b55b1abcf@10.0.0.1:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1003l stamp 30942
Content-Length: 233

v=0
o=- 1 2 IN IP4 10.0.0.5
s=CounterPath eyeBeam 1.5
c=IN IP4 10.0.0.5
t=0 0
m=audio 21966 RTP/AVP 0 8 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:106EB16556AA4E68BB3EB7029E9694C7
<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.5:21966
list_route: hop: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>
set_destination: Parsing <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b> for address/port to send to
set_destination: set destination to 10.0.0.5:8762
Transmitting (no NAT) to 10.0.0.5:8762:
ACK sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK00a0bd20
Max-Forwards: 70
From: "asterisk" <sip:asterisk@10.0.0.1>;tag=as09f98f70
To: <sip:102@10.0.0.5:8762;rinstance=12cb55d453468b0b>;tag=da5c7871
Contact: <sip:asterisk@10.0.0.1:5060>
Call-ID: 777b7a812c3fa1907620794b55b1abcf@10.0.0.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.0
Content-Length: 0


---
    -- SIP/102-0000437c answered DAHDI/7-1
  == Begin MixMonitor Recording DAHDI/7-1

localhost*CLI> rtp set debug 
off  on   ip   
localhost*CLI> rtp set debug on
RTP Debugging Enabled
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043673, ts 280800, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008698, ts 2884100, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043674, ts 280960, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008699, ts 2884260, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043675, ts 281120, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008700, ts 2884420, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043676, ts 281280, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008701, ts 2884580, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043677, ts 281440, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008702, ts 2884740, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043678, ts 281600, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043679, ts 281760, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008703, ts 2884900, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043680, ts 281920, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008704, ts 2885060, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008705, ts 2885220, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043681, ts 282080, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008706, ts 2885380, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043682, ts 282240, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008707, ts 2885540, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043683, ts 282400, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008708, ts 2885700, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043684, ts 282560, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008709, ts 2885860, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043685, ts 282720, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008710, ts 2886020, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043686, ts 282880, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008711, ts 2886180, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043687, ts 283040, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008712, ts 2886340, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043688, ts 283200, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008713, ts 2886500, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043689, ts 283360, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008714, ts 2886660, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043690, ts 283520, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008715, ts 2886820, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043691, ts 283680, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008716, ts 2886980, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043692, ts 283840, len 000160)
Got  RTP packet from    10.0.0.5:21966 (type 00, seq 008717, ts 2887140, len 000160)
Sent RTP packet to      10.0.0.5:21966 (type 00, seq 043693, ts 284000, len 000160)

Answer 1

1

Если на астере нет бульканья, проблема не в астере, очевидно, что при прочих равных, если х-лайт работает а самописный не работает, то проблема в самописном клиенте. Для выявления источника проблем используйте WireShark, покажет как все выглядит на уровне сети. Тут поищите много чего есть по данному вопросу.

ссылка удалить спам редактировать

ответил Apr 9 '12

itprofit
768 ● 24 ● 3 ● 27
http://itprofit32.ru/

Comments

Короче. Wireshark показал, что при звонке на самописный клиент, на принимающей стороне (GSM) 42% пакетов Dropby Jetter Buff и еще 14% пакетов имеют Wrong Timestamp. Кроме того, анализатор показал, что используется кодек G711, хотя в sip.conf и на клиенте принудительно оставлены только alaw, ulaw и gsm. Возникло 2 вопроса:

По основной проблеме я правильно понимаю, что нужно включать jitterbufer на asterisk?
Почему используется кодек G711?

foxm (Apr 13 '12)edit

alaw, ulaw это и есть g711 с разными алгоритмами кодирования, точнее это и есть алгоритмы.

alaw = g711a = pcma

alaw = g711u = pcmu

ro (Apr 13 '12)edit

Вам же написали, что проблема в самописном клиенте. И правильно написали.

ro (Apr 13 '12)edit

да. проблема решится если сделать forcejitterbuffer=yes. но правильно исправить клиент. а может и не решиться.если клеинт их в другом порядке передает.

meral (Apr 13 '12)edit

При таком раскладе, скорее всего, буфер дрожания не поможет, у вас наверно дельта ~600 джиттер под 300, в любом случае надо добиться нормальной работы клиента, иначе не по джидайски. Но может проблема вообще в сети попробуйте свой клиент на той машине запустить, где х-лайт работает нормально. А кодек в локалке использует 711, GSM вам зачем? Шлюзы, я так понял, у вас по аналоговой линии подключены?

itprofit (Apr 13 '12)edit

вобщето буфер помогает даже через спутник с 1200+ задержкой если что.

meral (Apr 13 '12)edit

можно и пол-часа буфер сделать, тогда можно будет RTP на флешке носить между компами. Задержка в секунду уже заметно осложняет общение. лечение буфером - крайний случай ИМХО

itprofit (Apr 13 '12)edit

Проблемы с голосом

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