[asterisk-bugs] [JIRA] (ASTERISK-25656) No Codec in 183 with Chan SIP and Asterisk 13 when Recoding Required
Peter Sokolov (JIRA)
noreply at issues.asterisk.org
Sun Jan 3 15:32:32 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25656?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Peter Sokolov updated ASTERISK-25656:
-------------------------------------
Description:
When the only way to establish the communication between two SIP devices is recoding, the 183 reply does not contain any sound codecs, causing the calling device to cancel the call.
In the following example the originating SIP device has only configured allow=g722 in sip.conf. The destination device has configured allow=g722,alaw, but replies with 183 containing alaw only. Asterisk 13.7.0-rc2 sends the 183 to the originating device, indicating no voice codecs at all, although it is capable to translate from alaw to g722. The originating device cancels the call. No "Adding codec" is visible in the SIP log between receiving the 183 and sending 183.
{code}
<--- SIP read from UDP:X.X.X.X:5060 --->
INVITE sip:123 at test.tst SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE
From: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
To: <sip:123 at test.tst>
Call-ID: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 INVITE
Contact: <sip:333 at X.X.X.X:5060;uniq=C465CA211E8EEE061CD248D47C840>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box 7490 113.06.50 (Dec 4 2015)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 431
v=0
o=user 16478182 16478182 IN IP4 X.X.X.X
s=call
c=IN IP4 X.X.X.X
t=0 0
m=audio 7078 RTP/AVP 9 8 0 2 102 100 99 101 97 120 121
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:120 PCMA/16000
a=rtpmap:121 PCMU/16000
a=rtcp:7079
a=ptime:30
<------------->
--- (18 headers 19 lines) ---
Sending to X.X.X.X:5060 (no NAT)
Using INVITE request as basis request - 2C3F4CB345A924C9 at 192.168.0.30
Found peer '333' for '333' from X.X.X.X:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 101
Found RTP audio format 97
Found RTP audio format 120
Found RTP audio format 121
Found audio description format G726-32 for ID 2
Found audio description format G726-32 for ID 102
Found unknown media description format G726-40 for ID 100
Found unknown media description format G726-24 for ID 99
Found audio description format telephone-event for ID 101
Found audio description format iLBC for ID 97
Found unknown media description format PCMA for ID 120
Found unknown media description format PCMU for ID 121
Capabilities: us - (g722), peer - audio=(ulaw|g726|alaw|g722|ilbc)/video=(nothing)/text=(nothing), combined - (g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port X.X.X.X:7078
Looking for 123 in users (domain test.tst)
sip_route_dump: route/path hop: <sip:333 at X.X.X.X:5060;uniq=C465CA211E8EEE061CD248D47C840>
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE;received=X.X.X.X
f: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
t: <sip:123 at test.tst>
i: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
m: <sip:123 at A.A.A.A:5060>
l: 0
<------------>
-- Executing [123 at users:1] NoOp("SIP/333-000000af", "") in new stack
-- Executing [123 at users:2] Set("SIP/333-000000af", "CALLERID(number)=00333") in new stack
-- Executing [123 at users:3] Set("SIP/333-000000af", "CALLERID(name)=333") in new stack
-- Executing [123 at users:4] Set("SIP/333-000000af", "CALLERID(number)=333") in new stack
-- Executing [123 at users:5] Dial("SIP/333-000000af", "SIP/123 at OutTrunk,60,") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17520
Adding codec g722 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to D.D.D.D:5060:
INVITE sip:123 at test2.tst SIP/2.0
v: SIP/2.0/UDP A.A.A.A:5060;branch=z9hG4bK57ab5e3d
Max-Forwards: 70
f: "333" <sip:333 at test2.tst>;tag=as0b8f7499
t: <sip:123 at test2.tst>
m: <sip:333 at A.A.A.A:5060>
i: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Sun, 03 Jan 2016 17:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
c: application/sdp
l: 278
v=0
o=root 1839661651 1839661651 IN IP4 X.X.X.X
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 X.X.X.X
t=0 0
m=audio 7078 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
-- Called SIP/123 at OutTrunk
<--- SIP read from UDP:D.D.D.D:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK57ab5e3d
From: "333" <sip:333 at test2.tst>;tag=as0b8f7499
To: <sip:123 at test2.tst>
Contact: sip:123 at D.D.D.D:5060
Call-ID: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:D.D.D.D:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK57ab5e3d
From: "333" <sip:333 at test2.tst>;tag=as0b8f7499
To: <sip:123 at test2.tst>;tag=3a0313ac55f03691abedc0
Contact: sip:123 at D.D.D.D:5060
Call-ID: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 210
v=0
o=CARRIER 1451841145 1451841145 IN IP4 D.D.D.D
s=SIP Call
c=IN IP4 D.D.D.D
t=0 0
m=audio 36114 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
sip_route_dump: route/path hop: <sip:0000038659207726 at D.D.D.D:5060>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port D.D.D.D:36114
-- SIP/OutTrunk-000000b0 is making progress passing it to SIP/333-000000af
Audio is at 10768
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE;received=X.X.X.X
f: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
t: <sip:123 at test.tst>;tag=as60d85cd4
i: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
m: <sip:123 at A.A.A.A:5060>
c: application/sdp
l: 203
v=0
o=root 2018919976 2018919976 IN IP4 D.D.D.D
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 D.D.D.D
t=0 0
m=audio 36114 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------>
<--- SIP read from UDP:X.X.X.X:5060 --->
CANCEL sip:123 at test.tst SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE
From: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
To: <sip:123 at test.tst>
Call-ID: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 CANCEL
Reason: SIP; cause=488; text="SDP problem"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box 7490 113.06.50 (Dec 4 2015)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0
{code}
Modifying the codecs to allow=g722,alaw for the originating device in sip.conf causes Asterisk to indicate both codecs (g722 and alaw) in 183 and it happily translates received alaw early media to g722 and sends it to the originating device. Two "Adding codec" are visible in the SIP log between receiving the 183 and sending 183.
{code}
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port D.D.D.D:11742
-- SIP/OutTrunk-000000b0 is making progress passing it to SIP/333-000000af
Audio is at 11122
Adding codec g722 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK4BDDDB221D20FE39;received=X.X.X.X
f: <sip:333 at test.tst>;tag=804A2A9B6FD4EF6D
t: <sip:123 at test.tst>;tag=as01c742c2
i: 6ED3E4C17715F952 at 192.168.0.30
CSeq: 3394 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
m: <sip:123 at A.A.A.A:5060>
c: application/sdp
l: 203
v=0
o=root 2040737164 2040737164 IN IP4 78.47.177.42
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 D.D.D.D
t=0 0
m=audio 36114 RTP/AVP 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
{code}
was:
When the only way to establish the communication between two SIP devices is recoding, the 183 reply does not contain any sound codecs, causing the calling device to cancel the call.
In the following example the originating SIP device has only configured allow=g722 in sip.conf. The destination device has configured allow=g722,alaw, but replies with 183 containing alaw only. Asterisk 13.7.0-rc2 sends the 183 to the originating device, indicating no voice codecs at all, although it is capable to translate from alaw to g722. The originating device cancels the call.
Modifying the codecs to allow=g722,alaw for the originating device in sip.conf causes Asterisk to indicate both codecs (g722 and alaw) in 183 and it happily translates received alaw early media to g722 and sends it to the originating device.
{code}
<--- SIP read from UDP:X.X.X.X:5060 --->
INVITE sip:123 at test.tst SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE
From: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
To: <sip:123 at test.tst>
Call-ID: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 INVITE
Contact: <sip:333 at X.X.X.X:5060;uniq=C465CA211E8EEE061CD248D47C840>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box 7490 113.06.50 (Dec 4 2015)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 431
v=0
o=user 16478182 16478182 IN IP4 X.X.X.X
s=call
c=IN IP4 X.X.X.X
t=0 0
m=audio 7078 RTP/AVP 9 8 0 2 102 100 99 101 97 120 121
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:120 PCMA/16000
a=rtpmap:121 PCMU/16000
a=rtcp:7079
a=ptime:30
<------------->
--- (18 headers 19 lines) ---
Sending to X.X.X.X:5060 (no NAT)
Using INVITE request as basis request - 2C3F4CB345A924C9 at 192.168.0.30
Found peer '333' for '333' from X.X.X.X:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 101
Found RTP audio format 97
Found RTP audio format 120
Found RTP audio format 121
Found audio description format G726-32 for ID 2
Found audio description format G726-32 for ID 102
Found unknown media description format G726-40 for ID 100
Found unknown media description format G726-24 for ID 99
Found audio description format telephone-event for ID 101
Found audio description format iLBC for ID 97
Found unknown media description format PCMA for ID 120
Found unknown media description format PCMU for ID 121
Capabilities: us - (g722), peer - audio=(ulaw|g726|alaw|g722|ilbc)/video=(nothing)/text=(nothing), combined - (g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port X.X.X.X:7078
Looking for 123 in users (domain test.tst)
sip_route_dump: route/path hop: <sip:333 at X.X.X.X:5060;uniq=C465CA211E8EEE061CD248D47C840>
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE;received=X.X.X.X
f: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
t: <sip:123 at test.tst>
i: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
m: <sip:123 at A.A.A.A:5060>
l: 0
<------------>
-- Executing [123 at users:1] NoOp("SIP/333-000000af", "") in new stack
-- Executing [123 at users:2] Set("SIP/333-000000af", "CALLERID(number)=00333") in new stack
-- Executing [123 at users:3] Set("SIP/333-000000af", "CALLERID(name)=333") in new stack
-- Executing [123 at users:4] Set("SIP/333-000000af", "CALLERID(number)=333") in new stack
-- Executing [123 at users:5] Dial("SIP/333-000000af", "SIP/123 at OutTrunk,60,") in new stack
== Using SIP RTP CoS mark 5
Audio is at 17520
Adding codec g722 to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to D.D.D.D:5060:
INVITE sip:123 at test2.tst SIP/2.0
v: SIP/2.0/UDP A.A.A.A:5060;branch=z9hG4bK57ab5e3d
Max-Forwards: 70
f: "333" <sip:333 at test2.tst>;tag=as0b8f7499
t: <sip:123 at test2.tst>
m: <sip:333 at A.A.A.A:5060>
i: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Sun, 03 Jan 2016 17:11:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
c: application/sdp
l: 278
v=0
o=root 1839661651 1839661651 IN IP4 X.X.X.X
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 X.X.X.X
t=0 0
m=audio 7078 RTP/AVP 9 8 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
-- Called SIP/123 at OutTrunk
<--- SIP read from UDP:D.D.D.D:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK57ab5e3d
From: "333" <sip:333 at test2.tst>;tag=as0b8f7499
To: <sip:123 at test2.tst>
Contact: sip:123 at D.D.D.D:5060
Call-ID: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:D.D.D.D:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK57ab5e3d
From: "333" <sip:333 at test2.tst>;tag=as0b8f7499
To: <sip:123 at test2.tst>;tag=3a0313ac55f03691abedc0
Contact: sip:123 at D.D.D.D:5060
Call-ID: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 210
v=0
o=CARRIER 1451841145 1451841145 IN IP4 D.D.D.D
s=SIP Call
c=IN IP4 D.D.D.D
t=0 0
m=audio 36114 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
sip_route_dump: route/path hop: <sip:0000038659207726 at D.D.D.D:5060>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port D.D.D.D:36114
-- SIP/OutTrunk-000000b0 is making progress passing it to SIP/333-000000af
Audio is at 10768
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 183 Session Progress
v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE;received=X.X.X.X
f: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
t: <sip:123 at test.tst>;tag=as60d85cd4
i: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
m: <sip:123 at A.A.A.A:5060>
c: application/sdp
l: 203
v=0
o=root 2018919976 2018919976 IN IP4 D.D.D.D
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 D.D.D.D
t=0 0
m=audio 36114 RTP/AVP 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------>
<--- SIP read from UDP:X.X.X.X:5060 --->
CANCEL sip:123 at test.tst SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE
From: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
To: <sip:123 at test.tst>
Call-ID: 2C3F4CB345A924C9 at 192.168.0.30
CSeq: 3394 CANCEL
Reason: SIP; cause=488; text="SDP problem"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box 7490 113.06.50 (Dec 4 2015)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0
{code}
> No Codec in 183 with Chan SIP and Asterisk 13 when Recoding Required
> --------------------------------------------------------------------
>
> Key: ASTERISK-25656
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25656
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Affects Versions: 13.7.0
> Environment: Debian 8.2
> Reporter: Peter Sokolov
>
> When the only way to establish the communication between two SIP devices is recoding, the 183 reply does not contain any sound codecs, causing the calling device to cancel the call.
> In the following example the originating SIP device has only configured allow=g722 in sip.conf. The destination device has configured allow=g722,alaw, but replies with 183 containing alaw only. Asterisk 13.7.0-rc2 sends the 183 to the originating device, indicating no voice codecs at all, although it is capable to translate from alaw to g722. The originating device cancels the call. No "Adding codec" is visible in the SIP log between receiving the 183 and sending 183.
> {code}
> <--- SIP read from UDP:X.X.X.X:5060 --->
> INVITE sip:123 at test.tst SIP/2.0
> Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE
> From: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
> To: <sip:123 at test.tst>
> Call-ID: 2C3F4CB345A924C9 at 192.168.0.30
> CSeq: 3394 INVITE
> Contact: <sip:333 at X.X.X.X:5060;uniq=C465CA211E8EEE061CD248D47C840>
> Max-Forwards: 70
> Expires: 120
> User-Agent: AVM FRITZ!Box 7490 113.06.50 (Dec 4 2015)
> Supported: 100rel,replaces
> Allow-Events: telephone-event,refer
> Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
> Content-Type: application/sdp
> Accept: application/sdp, multipart/mixed
> Accept-Encoding: identity
> Content-Length: 431
> v=0
> o=user 16478182 16478182 IN IP4 X.X.X.X
> s=call
> c=IN IP4 X.X.X.X
> t=0 0
> m=audio 7078 RTP/AVP 9 8 0 2 102 100 99 101 97 120 121
> a=sendrecv
> a=rtpmap:2 G726-32/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:100 G726-40/8000
> a=rtpmap:99 G726-24/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:120 PCMA/16000
> a=rtpmap:121 PCMU/16000
> a=rtcp:7079
> a=ptime:30
> <------------->
> --- (18 headers 19 lines) ---
> Sending to X.X.X.X:5060 (no NAT)
> Using INVITE request as basis request - 2C3F4CB345A924C9 at 192.168.0.30
> Found peer '333' for '333' from X.X.X.X:5060
> == Using SIP RTP CoS mark 5
> Found RTP audio format 9
> Found RTP audio format 8
> Found RTP audio format 0
> Found RTP audio format 2
> Found RTP audio format 102
> Found RTP audio format 100
> Found RTP audio format 99
> Found RTP audio format 101
> Found RTP audio format 97
> Found RTP audio format 120
> Found RTP audio format 121
> Found audio description format G726-32 for ID 2
> Found audio description format G726-32 for ID 102
> Found unknown media description format G726-40 for ID 100
> Found unknown media description format G726-24 for ID 99
> Found audio description format telephone-event for ID 101
> Found audio description format iLBC for ID 97
> Found unknown media description format PCMA for ID 120
> Found unknown media description format PCMU for ID 121
> Capabilities: us - (g722), peer - audio=(ulaw|g726|alaw|g722|ilbc)/video=(nothing)/text=(nothing), combined - (g722)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port X.X.X.X:7078
> Looking for 123 in users (domain test.tst)
> sip_route_dump: route/path hop: <sip:333 at X.X.X.X:5060;uniq=C465CA211E8EEE061CD248D47C840>
> <--- Transmitting (no NAT) to X.X.X.X:5060 --->
> SIP/2.0 100 Trying
> v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE;received=X.X.X.X
> f: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
> t: <sip:123 at test.tst>
> i: 2C3F4CB345A924C9 at 192.168.0.30
> CSeq: 3394 INVITE
> Server: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> k: replaces, timer
> m: <sip:123 at A.A.A.A:5060>
> l: 0
> <------------>
> -- Executing [123 at users:1] NoOp("SIP/333-000000af", "") in new stack
> -- Executing [123 at users:2] Set("SIP/333-000000af", "CALLERID(number)=00333") in new stack
> -- Executing [123 at users:3] Set("SIP/333-000000af", "CALLERID(name)=333") in new stack
> -- Executing [123 at users:4] Set("SIP/333-000000af", "CALLERID(number)=333") in new stack
> -- Executing [123 at users:5] Dial("SIP/333-000000af", "SIP/123 at OutTrunk,60,") in new stack
> == Using SIP RTP CoS mark 5
> Audio is at 17520
> Adding codec g722 to SDP
> Adding codec alaw to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to D.D.D.D:5060:
> INVITE sip:123 at test2.tst SIP/2.0
> v: SIP/2.0/UDP A.A.A.A:5060;branch=z9hG4bK57ab5e3d
> Max-Forwards: 70
> f: "333" <sip:333 at test2.tst>;tag=as0b8f7499
> t: <sip:123 at test2.tst>
> m: <sip:333 at A.A.A.A:5060>
> i: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
> CSeq: 102 INVITE
> User-Agent: Asterisk
> Date: Sun, 03 Jan 2016 17:11:35 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> k: replaces, timer
> c: application/sdp
> l: 278
> v=0
> o=root 1839661651 1839661651 IN IP4 X.X.X.X
> s=Asterisk PBX 13.7.0-rc2
> c=IN IP4 X.X.X.X
> t=0 0
> m=audio 7078 RTP/AVP 9 8 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> ---
> -- Called SIP/123 at OutTrunk
> <--- SIP read from UDP:D.D.D.D:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK57ab5e3d
> From: "333" <sip:333 at test2.tst>;tag=as0b8f7499
> To: <sip:123 at test2.tst>
> Contact: sip:123 at D.D.D.D:5060
> Call-ID: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
> CSeq: 102 INVITE
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Length: 0
> <------------->
> --- (10 headers 0 lines) ---
> <--- SIP read from UDP:D.D.D.D:5060 --->
> SIP/2.0 183 Session progress
> Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK57ab5e3d
> From: "333" <sip:333 at test2.tst>;tag=as0b8f7499
> To: <sip:123 at test2.tst>;tag=3a0313ac55f03691abedc0
> Contact: sip:123 at D.D.D.D:5060
> Call-ID: 7d2574bb50b5acc455fb91ff6859d6a9 at test2.tst
> CSeq: 102 INVITE
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Type: application/sdp
> Content-Length: 210
> v=0
> o=CARRIER 1451841145 1451841145 IN IP4 D.D.D.D
> s=SIP Call
> c=IN IP4 D.D.D.D
> t=0 0
> m=audio 36114 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=ptime:20
> a=sendrecv
> <------------->
> --- (11 headers 10 lines) ---
> sip_route_dump: route/path hop: <sip:0000038659207726 at D.D.D.D:5060>
> Found RTP audio format 8
> Found RTP audio format 101
> Found audio description format PCMA for ID 8
> Found audio description format telephone-event for ID 101
> Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port D.D.D.D:36114
> -- SIP/OutTrunk-000000b0 is making progress passing it to SIP/333-000000af
> Audio is at 10768
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Transmitting (no NAT) to X.X.X.X:5060 --->
> SIP/2.0 183 Session Progress
> v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE;received=X.X.X.X
> f: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
> t: <sip:123 at test.tst>;tag=as60d85cd4
> i: 2C3F4CB345A924C9 at 192.168.0.30
> CSeq: 3394 INVITE
> Server: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> k: replaces, timer
> m: <sip:123 at A.A.A.A:5060>
> c: application/sdp
> l: 203
> v=0
> o=root 2018919976 2018919976 IN IP4 D.D.D.D
> s=Asterisk PBX 13.7.0-rc2
> c=IN IP4 D.D.D.D
> t=0 0
> m=audio 36114 RTP/AVP 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=sendrecv
> <------------>
> <--- SIP read from UDP:X.X.X.X:5060 --->
> CANCEL sip:123 at test.tst SIP/2.0
> Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKB37C778E1ADC60AE
> From: <sip:333 at test.tst>;tag=8AF3B60254E4CDF1
> To: <sip:123 at test.tst>
> Call-ID: 2C3F4CB345A924C9 at 192.168.0.30
> CSeq: 3394 CANCEL
> Reason: SIP; cause=488; text="SDP problem"
> Max-Forwards: 70
> User-Agent: AVM FRITZ!Box 7490 113.06.50 (Dec 4 2015)
> Supported: 100rel,replaces
> Allow-Events: telephone-event,refer
> Content-Length: 0
> {code}
> Modifying the codecs to allow=g722,alaw for the originating device in sip.conf causes Asterisk to indicate both codecs (g722 and alaw) in 183 and it happily translates received alaw early media to g722 and sends it to the originating device. Two "Adding codec" are visible in the SIP log between receiving the 183 and sending 183.
> {code}
> Found RTP audio format 8
> Found RTP audio format 101
> Found audio description format PCMA for ID 8
> Found audio description format telephone-event for ID 101
> Capabilities: us - (g722|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port D.D.D.D:11742
> -- SIP/OutTrunk-000000b0 is making progress passing it to SIP/333-000000af
> Audio is at 11122
> Adding codec g722 to SDP
> Adding codec alaw to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Transmitting (no NAT) to X.X.X.X:5060 --->
> SIP/2.0 183 Session Progress
> v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK4BDDDB221D20FE39;received=X.X.X.X
> f: <sip:333 at test.tst>;tag=804A2A9B6FD4EF6D
> t: <sip:123 at test.tst>;tag=as01c742c2
> i: 6ED3E4C17715F952 at 192.168.0.30
> CSeq: 3394 INVITE
> Server: Asterisk
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> k: replaces, timer
> m: <sip:123 at A.A.A.A:5060>
> c: application/sdp
> l: 203
> v=0
> o=root 2040737164 2040737164 IN IP4 78.47.177.42
> s=Asterisk PBX 13.7.0-rc2
> c=IN IP4 D.D.D.D
> t=0 0
> m=audio 36114 RTP/AVP 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> {code}
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