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Call failed - Codec issue?


global_s
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Hi,

I have a weird PBX I presume. Sometimes, the two parties cannot hear each other. This happened with 3 different operators. I attached the full log but I think that the issue lies in these few lines

 

[6] 20120927110624: Call-leg 56: Codec pcmu/8000 is chosen for call id 3c28dfe898fe-85hxorlij0ms

[5] 20120927110624: set codec: codec pcmu/8000 is set to call-leg 56

[6] 20120927110633: Call port 56: Different Codecs (local , remote pcmu/8000), callid 172718c4@pbx, falling back to transcoding

 

Lock codec during conversation in on OFF. Previouly it was on and the last line was like, still no audio being sent

Call port 32: Different Codecs (local telephone-event/8000 , remote pcmu/8000), callid 172718c4@pbx, falling back to transcoding

 

Please help

 

INVITE sip:55922551131@sipmsk.dominus.com;phone=yes SIP/2.0
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-lr47idw7hno7;rport
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:50218@10.246.0.112:2064;transport=tls;line=spr0c3mz>;reg-id=1
X-Serialnumber: 00041337BDC5
P-Key-Flags: keys="3"
User-Agent: snom300/8.4.32
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Supported: timer, 100rel, replaces, from-change
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Require: buttons
Content-Type: application/sdp
Content-Length: 524

v=0
o=root 1066489614 1066489614 IN IP4 10.246.0.112
s=call
c=IN IP4 10.246.0.112
t=0 0
m=audio 65272 RTP/AVP 9 0 8 2 3 18 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:p/CVBq4J2wGZ4NFbSiOV1TnY43XJASftSISThwS1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt
a=sendrecv

[8] 20120927110624: Packet authenticated by transport layer
[8] 20120927110624: Allocating for call port 56, SIP call id 3c28dfe898fe-85hxorlij0ms 
[9] 20120927110624: UDP(IPv4): Opening socket on 0.0.0.0:10362
[9] 20120927110624: UDP(IPv4): Opening socket on 0.0.0.0:10363
[9] 20120927110624: UDP(IPv6): Opening socket on [::]:10362
[9] 20120927110624: UDP(IPv6): Opening socket on [::]:10363
[8] 20120927110624: Could not find a trunk (10 trunks)
[9] 20120927110624: Using outbound proxy sip:10.246.0.112:2064;transport=tls because of flow-label
[9] 20120927110624: Last message repeated 3 times
[5] 20120927110624: SIP Tx tls:10.246.0.112:2064:
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-lr47idw7hno7;rport=2064
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>;tag=c7a38b90bd
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 1 INVITE
Content-Length: 0


[7] 20120927110624: Set packet length to 20
[6] 20120927110624: Call-leg 56: Sending RTP for 3c28dfe898fe-85hxorlij0ms to 10.246.0.112:65272, codec not set yet
[8] 20120927110624: Incoming call: Request URI sip:55922551131@sipmsk.dominus.com;phone=yes, To is <sip:55922551131@sipmsk.dominus.com;phone=yes>
[8] 20120927110624: Call from an user 50218
[8] 20120927110624: To is <sip:55922551131@sipmsk.dominus.com;phone=yes>, user 0, domain 2
[8] 20120927110624: From user 50218
[8] 20120927110624: Set the To domain based on From user 50218@sipmsk.dominus.com
[8] 20120927110624: Call state for call object 748: idle
[9] 20120927110624: Dialplan: Evaluating !^(\+?[0-9]*)@.*!sip:00\1@\r;user=phone!i against 55922551131@sipmsk.dominus.com
[7] 20120927110624: Call port 56: set_codecs for 3c28dfe898fe-85hxorlij0ms codecs "", codec_preference count 7
[5] 20120927110624: Dialplan "Hiden CallerID": Match 55922551131@sipmsk.dominus.com to sip:0055922551131@sip.trtan.com;user=phone on trunk trtan
[9] 20120927110624: Generating hf header using {trunk}
[9] 20120927110624: Generating ht header using {request-uri}
[8] 20120927110624: Play audio_moh/noise.wav, caching true
[8] 20120927110624: Allocating for call port 57, SIP call id 172718c4@pbx 
[9] 20120927110624: UDP(IPv4): Opening socket on 0.0.0.0:10666
[9] 20120927110624: UDP(IPv4): Opening socket on 0.0.0.0:10667
[9] 20120927110624: UDP(IPv6): Opening socket on [::]:10666
[9] 20120927110624: UDP(IPv6): Opening socket on [::]:10667
[7] 20120927110624: Call port 57: set_codecs for 172718c4@pbx codecs "", codec_preference count 7
[8] 20120927110624: call port 57: state code from 0 to 100
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: adding codec pcmu/8000 to available list
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: adding codec pcma/8000 to available list
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: adding codec g722/8000 to available list
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: adding codec g729/8000 to available list
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: adding codec g726-32/8000 to available list
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: adding codec gsm/8000 to available list
[9] 20120927110624: Call port 57: update_codecs for 172718c4@pbx: codec_preference size 7, available codecs size 7
[9] 20120927110624: Resolve 82775: url sip:sip.trtan.com:5060
[9] 20120927110624: Resolve 82775: a udp sip.trtan.com 5060
[8] 20120927110624: DNS: Request sip.trtan.com from server 8.8.8.8
[8] 20120927110624: call port 56: state code from 0 to 183
[7] 20120927110624: Set packet length to 20
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: adding codec pcmu/8000 to available list
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: adding codec pcma/8000 to available list
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: adding codec g722/8000 to available list
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: adding codec g729/8000 to available list
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: adding codec g726-32/8000 to available list
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: adding codec gsm/8000 to available list
[9] 20120927110624: Call port 56: update_codecs for 3c28dfe898fe-85hxorlij0ms: codec_preference size 7, available codecs size 7
[6] 20120927110624: Call-leg 56: Codec pcmu/8000 is chosen for call id 3c28dfe898fe-85hxorlij0ms
[5] 20120927110624: set codec: codec pcmu/8000 is set to call-leg 56
[5] 20120927110624: SIP Tx tls:10.246.0.112:2064:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-lr47idw7hno7;rport=2064
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>;tag=c7a38b90bd
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 1 INVITE
Contact: <sip:50218@10.252.0.8:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Require: 100rel
RSeq: 1
Content-Type: application/sdp
Content-Length: 474

v=0
o=- 768700253 768700253 IN IP4 10.252.0.8
s=-
c=IN IP4 10.252.0.8
t=0 0
m=audio 10362 RTP/AVP 0 8 9 18 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:LBmzkoeE9loWtfUu9juWsDcRR8y0Q170IPEy7vKG
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[8] 20120927110624: DNS: Add A sip.trtan.com 83.123.111.99 (ttl=277)
[9] 20120927110624: DNS: erasing A sip.trtan.com, id 290 retry count 0,
[9] 20120927110624: Resolve 82775: a udp sip.trtan.com 5060
[9] 20120927110624: Resolve 82775: udp 83.123.111.99 5060
[5] 20120927110624: SIP Tx udp:83.123.111.99:5060:
INVITE sip:0055922551131@sip.trtan.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-0235eb9178040faccd38311dc0fdcd08;rport
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>
Call-ID: 172718c4@pbx
CSeq: 10834 INVITE
Max-Forwards: 70
Contact: <sip:99051000170744@5.923.102.70:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
P-Charging-Vector: icid-value=;icid-generated-at=5.923.102.70;orig-ioi=sipmsk.dominus.com
Content-Type: application/sdp
Content-Length: 382

v=0
o=- 813988938 813988938 IN IP4 5.923.102.70
s=-
c=IN IP4 5.923.102.70
t=0 0
m=audio 10666 RTP/AVP 0 8 9 18 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[6] 20120927110624: Received bindRequest for user sipmsk.dominus.com\50218
[5] 20120927110624: SIP Rx tls:10.246.0.112:2064:
PRACK sip:50218@10.252.0.8:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-g3kknk87lo3v;rport
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>;tag=c7a38b90bd
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 2 PRACK
Max-Forwards: 70
Contact: <sip:50218@10.246.0.112:2064;transport=tls;line=spr0c3mz>;reg-id=1
RAck: 1 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE
Allow-Events: talk, hold, refer, call-info
Proxy-Require: buttons
Content-Length: 0


[8] 20120927110624: Packet authenticated by transport layer
[5] 20120927110624: SIP Tx tls:10.246.0.112:2064:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-g3kknk87lo3v;rport=2064
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>;tag=c7a38b90bd
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 2 PRACK
Contact: <sip:50218@10.252.0.8:5061;transport=tls>
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Length: 0


[5] 20120927110624: SIP Rx udp:83.123.111.99:5060:
SIP/2.0 100 Trying
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>
Call-ID: 172718c4@pbx
CSeq: 10834 INVITE
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-0235eb9178040faccd38311dc0fdcd08;rport=5060
Contact: <sip:0055922551131@sip.trtan.com:5060;user=phone;maddr=83.123.111.99;transport=udp>
Content-Length: 0


[9] 20120927110624: Message repetition, packet dropped
[8] 20120927110624: SRTP MAC mismatch: e09b3222 != 4f4d0000
[7] 20120927110624: Discard SRTCP packet from 10.246.0.112:65273 with wrong MAC
[6] 20120927110624: Received searchRequest, equalityMatch (description=telephoneNumber, value=+22551131)
[6] 20120927110624: Received searchRequest(type 128), substrings=0022551131
[5] 20120927110624: SIP Rx udp:83.123.111.99:5060:
SIP/2.0 407 Proxy Authentication Required
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>;tag=c990d13f-13c4-50647a35-af39778-714f4fbd
Call-ID: 172718c4@pbx
CSeq: 10834 INVITE
Proxy-Authenticate: Digest realm="sip.trtan.com", nonce="50647a520000d44ff87226a0d7d58745398fa8616e1632ad", algorithm=MD5
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-0235eb9178040faccd38311dc0fdcd08;rport=5060
Content-Length: 0


[8] 20120927110624: Answer challenge with username 99051000170744
[9] 20120927110624: Resolve 82778: udp 83.123.111.99 5060 udp:1
[9] 20120927110624: Resolve 82779: udp 83.123.111.99 5060 udp:1
[5] 20120927110624: SIP Tx udp:83.123.111.99:5060:
INVITE sip:0055922551131@sip.trtan.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-e8c52184a380bee58cf4f63daff5ffd1;rport
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>
Call-ID: 172718c4@pbx
CSeq: 10835 INVITE
Max-Forwards: 70
Contact: <sip:99051000170744@5.923.102.70:5060;transport=udp>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
P-Charging-Vector: icid-value=;icid-generated-at=5.923.102.70;orig-ioi=sipmsk.dominus.com
Proxy-Authorization: Digest realm="sip.trtan.com",nonce="50647a520000d44ff87226a0d7d58745398fa8616e1632ad",response="816504d42f693bb952702e711ff49535",username="99051000170744",uri="sip:0055922551131@sip.trtan.com;user=phone",algorithm=MD5
Content-Type: application/sdp
Content-Length: 382

v=0
o=- 813988938 813988938 IN IP4 5.923.102.70
s=-
c=IN IP4 5.923.102.70
t=0 0
m=audio 10666 RTP/AVP 0 8 9 18 2 3 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[9] 20120927110624: Message repetition, packet dropped
[5] 20120927110624: SIP Rx udp:83.123.111.99:5060:
SIP/2.0 100 Trying
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>;tag=c990d13f-13c4-50647a35-af39778-714f4fbd
Call-ID: 172718c4@pbx
CSeq: 10835 INVITE
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-e8c52184a380bee58cf4f63daff5ffd1;rport=5060
Contact: <sip:0055922551131@sip.trtan.com:5060;user=phone;maddr=83.123.111.99;transport=udp>
Content-Length: 0


[9] 20120927110624: Message repetition, packet dropped
[8] 20120927110626: Packet authenticated by transport layer
[9] 20120927110626: Resolve 82781: udp 82.112.213.51 5060
[5] 20120927110626: SIP Rx udp:83.123.111.99:5060:
SIP/2.0 180 Ringing
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>;tag=c990d13f-13c4-50647a35-af39778-714f4fbd
Call-ID: 172718c4@pbx
CSeq: 10835 INVITE
User-Agent: SipGW 9
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-e8c52184a380bee58cf4f63daff5ffd1;rport=5060
Contact: <sip:0055922551131@sip.trtan.com:5060;user=phone;maddr=83.123.111.99;transport=udp>
Content-Length: 0


[8] 20120927110626: Call state for call object 748: alerting
[8] 20120927110626: Play audio_it/ringback.wav, caching true
[8] 20120927110626: call port 56: state code from 183 to 183
[9] 20120927110627: Resolve 82782: udp 82.112.213.51 5060
[9] 20120927110627: Resolve 82783: aaaa udp 188.40.65.170 27010
[9] 20120927110627: Resolve 82783: a udp 188.40.65.170 27010
[9] 20120927110627: Resolve 82783: udp 188.40.65.170 27010
[8] 20120927110627: Packet authenticated by transport layer
[6] 20120927110629: Call-leg 57: Sending RTP for 172718c4@pbx to 83.123.111.99:24502, codec not set yet
[5] 20120927110630: SIP Rx udp:83.123.111.99:5060:
SIP/2.0 200 OK
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>;tag=c990d13f-13c4-50647a35-af39778-714f4fbd
Call-ID: 172718c4@pbx
CSeq: 10835 INVITE
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE
User-Agent: SipGW 9
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-e8c52184a380bee58cf4f63daff5ffd1;rport=5060
Contact: <sip:0055922551131@sip.trtan.com:5060;user=phone;maddr=83.123.111.99;transport=udp>
Content-Type: application/sdp
Content-Length: 184

v=0
o=- 813988938 813988938 IN IP4 83.123.111.99
s=trtan call
c=IN IP4 83.123.111.99
t=0 0
m=audio 24502 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

[7] 20120927110630: Call 172718c4@pbx: Clear last INVITE
[6] 20120927110630: Call-leg 57: Codec pcmu/8000 is chosen for call id 172718c4@pbx
[5] 20120927110630: set codec: codec pcmu/8000 is set to call-leg 57
[9] 20120927110630: Resolve 82785: aaaa udp 83.123.111.99 5060
[9] 20120927110630: Resolve 82785: a udp 83.123.111.99 5060
[9] 20120927110630: Resolve 82785: udp 83.123.111.99 5060
[5] 20120927110630: SIP Tx udp:83.123.111.99:5060:
ACK sip:0055922551131@sip.trtan.com:5060;user=phone;maddr=83.123.111.99;transport=udp SIP/2.0
Via: SIP/2.0/UDP 5.923.102.70:5060;branch=z9hG4bK-4ed57a8ab20538ab1658161d80db4ea6;rport
From: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
To: <sip:0055922551131@sip.trtan.com>;tag=c990d13f-13c4-50647a35-af39778-714f4fbd
Call-ID: 172718c4@pbx
CSeq: 10835 ACK
Max-Forwards: 70
Contact: <sip:99051000170744@5.923.102.70:5060;transport=udp>
P-Charging-Vector: icid-value=;icid-generated-at=5.923.102.70;orig-ioi=sipmsk.dominus.com
Content-Length: 0


[7] 20120927110630: Determine pass-through mode after receiving response
[8] 20120927110630: Call state for call object 748: connected
[8] 20120927110630: call port 57: state code from 100 to 200
[8] 20120927110630: call port 56: state code from 183 to 200
[5] 20120927110630: SIP Tx tls:10.246.0.112:2064:
SIP/2.0 200 Ok
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-lr47idw7hno7;rport=2064
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>;tag=c7a38b90bd
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 1 INVITE
Contact: <sip:50218@10.252.0.8:5061;transport=tls>
Supported: 100rel, replaces, norefersub
Allow-Events: refer
Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE
Accept: application/sdp
User-Agent: snomONE/4.5.0.1075 Delta Aurigids
Content-Type: application/sdp
Content-Length: 474

v=0
o=- 768700253 768700253 IN IP4 10.252.0.8
s=-
c=IN IP4 10.252.0.8
t=0 0
m=audio 10362 RTP/AVP 0 8 9 18 2 3 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:LBmzkoeE9loWtfUu9juWsDcRR8y0Q170IPEy7vKG
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=rtcp-xr:rcvr-rtt=all voip-metrics
a=sendrecv

[7] 20120927110630: 3c28dfe898fe-85hxorlij0ms: RTP pass-through mode
[7] 20120927110630: 172718c4@pbx: RTP pass-through mode
[5] 20120927110630: SIP Rx tls:10.246.0.112:2064:
ACK sip:50218@10.252.0.8:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 10.246.0.112:2064;branch=z9hG4bK-1xvgmo1rz5i1;rport
From: "helo Group" <sip:50218@sipmsk.dominus.com>;tag=o3t5y4nlpc
To: <sip:55922551131@sipmsk.dominus.com;phone=yes>;tag=c7a38b90bd
Call-ID: 3c28dfe898fe-85hxorlij0ms
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:50218@10.246.0.112:2064;transport=tls;line=spr0c3mz>;reg-id=1
Proxy-Require: buttons
Content-Length: 0


[8] 20120927110630: Packet authenticated by transport layer
[8] 20120927110633: Last message repeated 3 times
[6] 20120927110633: Call port 56: Different Codecs (local , remote pcmu/8000), callid 172718c4@pbx, falling back to transcoding
[8] 20120927110639: Packet authenticated by transport layer
[8] 20120927110641: Last message repeated 2 times
[5] 20120927110641: SIP Rx udp:83.123.111.99:5060:
BYE sip:99051000170744@5.923.102.70:5060;transport=udp SIP/2.0
From: <sip:0055922551131@sip.trtan.com>;tag=c990d13f-13c4-50647a35-af39778-714f4fbd
To: "99051000170744" <sip:99051000170744@sip.trtan.com>;tag=964326343
Call-ID: 172718c4@pbx
CSeq: 1 BYE
Via: SIP/2.0/UDP 83.123.111.99:5060;branch=z9hG4bK-cb68f-50647a46-af3da64-18779120
Max-Forwards: 70
Content-Length: 0

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you mean on the pbx or on the phone?

 

On the phone it is not going to be possible as it is in production. The weird thing is that it only happens sometimes, not always.

 

Do you think it is a good idea to tell SnomOne to only use pcmu 8000 as available codec?

 

Hellp, this issue is really making my customers unhappy. What I could read in the log I posted is that somehow, the local party decide to switch codec, or worse, SnomONE forget the local codec.

 

We are using snom300 with fw8.4.32 as local party with PnP.

 

Please help us

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The PCAP would be on the PBX.

 

Yes, I would for now limit the codec on the system to PCMU. If you have enough bandwidth, this is definitively a good idea. Then after hours, you can try to get a PCAP with the codecs unlimited to see where the problem is.

 

At first I tried to only restrict codec on the trunk side, but the issue was still there.

Now I changed the system setting to only allow G711U, and I'm waiting for time to pass, hoping that no other issues arise.

In case we found a fix, I will not able to do the troubleshooting you asked because it a pbx in production and I don't want to create problems on purpose.

 

Please understand my concerns.

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At first I tried to only restrict codec on the trunk side, but the issue was still there.

Now I changed the system setting to only allow G711U, and I'm waiting for time to pass, hoping that no other issues arise.

In case we found a fix, I will not able to do the troubleshooting you asked because it a pbx in production and I don't want to create problems on purpose.

Please understand my concerns.

 

Hello, I'm setting up another trunk and the issue is still there

 

[6] 2012/10/02 15:36:49: Call port 59: Different Codecs (local pcmu/8000, remote g729/8000), callid 3c2e5f2ba4c9-qnr3hgnzb9un, falling back to transcoding

[6] 2012/10/02 15:36:49: Call port 58: Different Codecs (local g729/8000, remote pcmu/8000), callid 329a68fd@pbx, falling back to transcoding

 

The problem is that I setup the system to only use G711...how can he accept g729?

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It depends who proposes the codec. If the carrier insists on G.729A, there is little that the PBX can do but accept it. Two things: Transcoding is not evil per se. The log entries above look okay to me, and at least both parties should be able to talk normally even though transcoding does reduce the audio quality. Second, there must be something special with that trunk. This is not a common problem. Maybe the provider takes the liberty to switch the codec without properly negotiating it; SIP is not very strict with such things. I know it is difficult to get a PCAP, but a PCAP would be able to pinpoint the problem.

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It depends who proposes the codec. If the carrier insists on G.729A, there is little that the PBX can do but accept it. Two things: Transcoding is not evil per se. The log entries above look okay to me, and at least both parties should be able to talk normally even though transcoding does reduce the audio quality. Second, there must be something special with that trunk. This is not a common problem. Maybe the provider takes the liberty to switch the codec without properly negotiating it; SIP is not very strict with such things. I know it is difficult to get a PCAP, but a PCAP would be able to pinpoint the problem.

 

Thank you for the reply.

I know transcoding it is not evil, but it simply doesn't work in my PBX. That is why I had to limit the codec.

 

I'm using now two providers; they have Codec override set to only accept g711U and the pbx is se to only accept G711U. Now my pbx works.

 

The messages "falling to trasncoding" resulted in one party not hearing. This is why I'm saying that transcoding is failing for me, and it's here that I need you help :)

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In this case we/you really have to look at the RTP streams (In/Out). If the streams are different, then on some devices you will see this issue. The "Lock codec" (at the Admin->Settings->Ports page) supposed to take care of this very issue, i.e., if the phone is doing codec A and if the trunk provider is doing codec B, PBX will transcode codec B to codec A and send it to the phone.

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