[asterisk-bugs] [JIRA] (ASTERISK-25676) native_rtp Lets through Codec not in the Allowed List instead of Transcoding
Peter Sokolov (JIRA)
noreply at issues.asterisk.org
Sat Jan 9 17:21:32 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25676?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Peter Sokolov updated ASTERISK-25676:
-------------------------------------
Description:
The call originating SIP device only has g722 in the list of allowed codecs and directmedia=no set in its configuration. On a call to SIP destination that replies with alaw (183 and OK), Asterisk 13 will correctly transcode alaw to g722 during the early media, however as soon as the connection is established (OK) and the bridge switches to native_rtp, it will simply let through the alaw packets.
To reproduce, configure like below. 333 is a SIP phone (FritzBox in my case) that supports g722 and alaw. OutTrunk is a SIP trunk provider that does not support g722 for calls out of its network (it replies with alaw only in 183 and OK SDP). At least one peer has to be set to directmedia=no in a way that RTP always flows via Asterisk.
Make a call from peer 333 to a number that is routed through OutTrunk. When OutTrunk sends 183 with only alaw in SDP, you will see that Asterisk sends 183 with g722 in SDP to peer 333 and will transcode the alaw early media to g722. As soon as the connection is established and OutTrunk sends OK with only alaw in SDP, you will see that Asterisk sends OK with g722 in SDP to peer 333, however native_rtp bridge will let through the alaw media to the peer 333 even though alaw is not in the list of allowed codecs for the peer 333.
{code}
sip.conf:
[333]
type=friend
context=users
subscribecontext=users
secret=donttell
language=de
mailbox=333 at default
host=dynamic
dtmfmode=rfc2833
defaultuser=333
callerid=333 <00333>
restrictcid=no
fromdomain=test.tst
directmedia=no
nat=no
insecure=port
call-limit=5
allowsubscribe=yes
disallow=all
allow=g722
[OutTrunk]
type=peer
fromdomain=test2.tst
defaultuser=user1
secret=donttell
host=test2.tst
context=users
directmedia=yes
qualify=no
insecure=port,invite
promiscredir=no
dtmfmode=rfc2833
nat=no
disallow=all
allow=g722,alaw
callgroup=1
t38pt_udptl=no
t38pt_rtp=no
t38pt_tcp=no
extensions.conf:
[users]
exten => 123,1,NoOp
exten => 123,n,Set(CALLERID(number)=00333)
exten => 123,n,Set(CALLERID(name)=${CALLERID(number):2})
exten => 123,n,Set(CALLERID(number)=${CALLERID(number):2})
exten => 123,n,Dial(SIP/${EXTEN}@OutTrunk,60,)
exten => 123,n,Hangup
{code}
Here is the log where "Sent RTP P2P packet" forwarding the alaw packet without transcoding can be seen:
{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=z9hG4bK7D8CF25D90C364B9
From: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
To: <sip:123 at test.tst>
Call-ID: 241FB38ED47A0E77 at 192.168.0.30
CSeq: 2573 INVITE
Contact: <sip:333 at X.X.X.X;uniq=C09040DC74C0F2A9D6039136CB478>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box 7330 107.06.30 (Jul 10 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: 433
v=0
o=user 9081636 9081636 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 97 120 121 101
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:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:120 PCMA/16000
a=rtpmap:121 PCMU/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
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 - 241FB38ED47A0E77 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 97
Found RTP audio format 120
Found RTP audio format 121
Found RTP audio format 101
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 iLBC for ID 97
Found unknown media description format PCMA for ID 120
Found unknown media description format PCMU for ID 121
Found audio description format telephone-event for ID 101
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;uniq=C09040DC74C0F2A9D6039136CB478>
<--- 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=z9hG4bK7D8CF25D90C364B9;received=X.X.X.X
f: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
t: <sip:123 at test.tst>
i: 241FB38ED47A0E77 at 192.168.0.30
CSeq: 2573 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/333-000001f3", "") in new stack
-- Executing [123 at users:2] NoOp("SIP/333-000001f3", "") in new stack
-- Executing [123 at users:3] Set("SIP/333-000001f3", "CALLERID(name)=333") in new stack
-- Executing [123 at users:4] Set("SIP/333-000001f3", "CALLERID(number)=333") in new stack
-- Executing [123 at users:5] Dial("SIP/333-000001f3", "SIP/123 at OutTrunk,60,") in new stack
== Using SIP RTP CoS mark 5
Audio is at 12608
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=z9hG4bK5b61c9e2
Max-Forwards: 70
f: "333" <sip:333 at test2.tst>;tag=as4c31698e
t: <sip:123 at test2.tst>
m: <sip:333 at A.A.A.A:5060>
i: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Sat, 09 Jan 2016 21:27:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
k: replaces, timer
c: application/sdp
l: 279
v=0
o=root 1956784559 1956784559 IN IP4 A.A.A.A
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 A.A.A.A
t=0 0
m=audio 12608 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=z9hG4bK5b61c9e2
From: "333" <sip:333 at test2.tst>;tag=as4c31698e
To: <sip:123 at test2.tst>
Contact: sip:123 at D.D.D.D:5060
Call-ID: 669e68ca24c091413a62b0e06a0c2128 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=z9hG4bK5b61c9e2
From: "333" <sip:333 at test2.tst>;tag=as4c31698e
To: <sip:123 at test2.tst>;tag=360313ac55ed93a8b99b71
Contact: sip:123 at D.D.D.D:5060
Call-ID: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 212
v=0
o=CARRIER 1452374900 1452374900 IN IP4 G.G.G.G
s=SIP Call
c=IN IP4 G.G.G.G
t=0 0
m=audio 26660 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:123 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 G.G.G.G:26660
-- SIP/OutTrunk-000001f4 is making progress passing it to SIP/333-000001f3
Audio is at 17508
Adding codec g722 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=z9hG4bK7D8CF25D90C364B9;received=X.X.X.X
f: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
t: <sip:123 at test.tst>;tag=as521bc73b
i: 241FB38ED47A0E77 at 192.168.0.30
CSeq: 2573 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: 253
v=0
o=root 486862468 486862468 IN IP4 A.A.A.A
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 A.A.A.A
t=0 0
m=audio 17508 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
> 0x27e1720 -- Probation passed - setting RTP source address to G.G.G.G:26660
Got RTP packet from G.G.G.G:26660 (type 08, seq 026726, ts 3738337846, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007407, ts 000120, len 000160)
Got RTP packet from G.G.G.G:26660 (type 08, seq 026727, ts 3738338006, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007408, ts 000280, len 000160)
Got RTP packet from G.G.G.G:26660 (type 08, seq 026728, ts 3738338166, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007409, ts 000440, len 000160)
Got RTP packet from G.G.G.G:26660 (type 08, seq 026729, ts 3738338326, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007410, ts 000600, len 000160)
Got RTP packet from G.G.G.G:26660 (type 08, seq 026730, ts 3738338486, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007411, ts 000760, len 000160)
> 0x253cd30 -- Probation passed - setting RTP source address to X.X.X.X:7078
Got RTP packet from X.X.X.X:7078 (type 09, seq 000001, ts 000240, len 000160)
Sent RTP packet to G.G.G.G:26660 (type 08, seq 049922, ts 000152, len 000160)
Got RTP packet from G.G.G.G:26660 (type 08, seq 026731, ts 3738338646, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007412, ts 000920, len 000160)
Got RTP packet from X.X.X.X:7078 (type 09, seq 000002, ts 000480, len 000240)
Sent RTP packet to G.G.G.G:26660 (type 08, seq 049923, ts 000392, len 000240)
Got RTP packet from G.G.G.G:26660 (type 08, seq 026732, ts 3738338806, len 000160)
Sent RTP packet to X.X.X.X:7078 (type 09, seq 007413, ts 001080, len 000160)
<--- SIP read from UDP:D.D.D.D:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK5b61c9e2
From: "333" <sip:333 at test2.tst>;tag=as4c31698e
To: <sip:123 at test2.tst>;tag=360313ac55ed93a8b99b71
Contact: sip:123 at D.D.D.D:5060
Call-ID: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
CSeq: 102 INVITE
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 212
v=0
o=CARRIER 1452374913 1452374913 IN IP4 G.G.G.G
s=SIP Call
c=IN IP4 G.G.G.G
t=0 0
m=audio 26660 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 10 lines) ---
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 G.G.G.G:26660
sip_route_dump: route/path hop: <sip:123 at D.D.D.D:5060>
set_destination: Parsing <sip:123 at D.D.D.D:5060> for address/port to send to
set_destination: set destination to D.D.D.D:5060
Transmitting (no NAT) to D.D.D.D:5060:
ACK sip:123 at D.D.D.D:5060 SIP/2.0
v: SIP/2.0/UDP A.A.A.A:5060;branch=z9hG4bK0442d057
Max-Forwards: 70
f: "333" <sip:333 at test2.tst>;tag=as4c31698e
t: <sip:123 at test2.tst>;tag=360313ac55ed93a8b99b71
m: <sip:333 at A.A.A.A:5060>
i: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
CSeq: 102 ACK
User-Agent: Asterisk
l: 0
---
-- SIP/OutTrunk-000001f4 answered SIP/333-000001f3
Audio is at 17508
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to X.X.X.X:5060 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK7D8CF25D90C364B9;received=X.X.X.X
f: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
t: <sip:123 at test.tst>;tag=as521bc73b
i: 241FB38ED47A0E77 at 192.168.0.30
CSeq: 2573 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: 253
v=0
o=root 486862468 486862468 IN IP4 A.A.A.A
s=Asterisk PBX 13.7.0-rc2
c=IN IP4 A.A.A.A
t=0 0
m=audio 17508 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
-- Channel SIP/OutTrunk-000001f4 joined 'simple_bridge' basic-bridge <94935aef-db90-4ff7-ac85-60d04663af6d>
-- Channel SIP/333-000001f3 joined 'simple_bridge' basic-bridge <94935aef-db90-4ff7-ac85-60d04663af6d>
> Bridge 94935aef-db90-4ff7-ac85-60d04663af6d: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/333-000001f3' and 'SIP/OutTrunk-000001f4' in stack
> Locally RTP bridged 'SIP/333-000001f3' and 'SIP/OutTrunk-000001f4' in stack
> 0x27e1720 -- Probation passed - setting RTP source address to G.G.G.G:26660
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
<--- SIP read from UDP:X.X.X.X:5060 --->
ACK sip:123 at 172.31.1.100:5060 SIP/2.0
Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKF1AA2AEE2EEA0175
From: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
To: <sip:123 at test.tst>;tag=as521bc73b
Call-ID: 241FB38ED47A0E77 at 192.168.0.30
CSeq: 2573 ACK
Contact: <sip:333 at X.X.X.X;uniq=C09040DC74C0F2A9D6039136CB478>
Max-Forwards: 70
User-Agent: AVM FRITZ!Box 7330 107.06.30 (Jul 10 2015)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Got RTP packet from X.X.X.X:7078 (type 09, seq 000409, ts 098160, len 000240)
Sent RTP packet to G.G.G.G:26660 (type 08, seq 050330, ts 098072, len 000240)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Got RTP packet from X.X.X.X:7078 (type 09, seq 000410, ts 098560, len 000400)
Sent RTP packet to G.G.G.G:26660 (type 08, seq 050331, ts 098472, len 000400)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
Got RTP packet from X.X.X.X:7078 (type 09, seq 000411, ts 098800, len 000240)
Sent RTP packet to G.G.G.G:26660 (type 08, seq 050332, ts 098712, len 000240)
Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
{code}
was:
The call originating SIP device only has g722 in the list of allowed codecs and directmedia=no set in its configuration. On a call to SIP destination that replies with alaw (183 and OK), Asterisk 13 will correctly transcode alaw to g722 during the early media, however as soon as the connection is established (OK) and the bridge switches to native_rtp, it will simply let through the alaw packets.
To reproduce, configure like below. 333 is a SIP phone (FritzBox in my case) that supports g722 and alaw. OutTrunk is a SIP trunk provider that does not support g722 for calls out of its network (it replies with alaw only in 183 and OK SDP). At least one peer has to be set to directmedia=no in a way that RTP always flows via Asterisk.
Make a call from peer 333 to a number that is routed through OutTrunk. When OutTrunk sends 183 with only alaw in SDP, you will see that Asterisk sends 183 with g722 in SDP to peer 333 and will transcode the alaw early media to g722. As soon as the connection is established and OutTrunk sends OK with only alaw in SDP, you will see that Asterisk sends OK with g722 in SDP to peer 333, however native_rtp bridge will let through the alaw media to the peer 333 even though alaw is not in the list of allowed codecs for the peer 333.
{code}
sip.conf:
[333]
type=friend
context=users
subscribecontext=users
secret=donttell
language=de
mailbox=333 at default
host=dynamic
dtmfmode=rfc2833
defaultuser=333
callerid=333 <00333>
restrictcid=no
fromdomain=test.tst
directmedia=no
nat=no
insecure=port
call-limit=5
allowsubscribe=yes
disallow=all
allow=g722
[OutTrunk]
type=peer
fromdomain=test2.tst
defaultuser=user1
secret=donttell
host=test2.tst
context=users
directmedia=yes
qualify=no
insecure=port,invite
promiscredir=no
dtmfmode=rfc2833
nat=no
disallow=all
allow=g722,alaw
callgroup=1
t38pt_udptl=no
t38pt_rtp=no
t38pt_tcp=no
extensions.conf:
[users]
exten => 123,1,NoOp
exten => 123,n,Set(CALLERID(number)=00333)
exten => 123,n,Set(CALLERID(name)=${CALLERID(number):2})
exten => 123,n,Set(CALLERID(number)=${CALLERID(number):2})
exten => 123,n,Dial(SIP/${EXTEN}@OutTrunk,60,)
exten => 123,n,Hangup
{code}
> native_rtp Lets through Codec not in the Allowed List instead of Transcoding
> ----------------------------------------------------------------------------
>
> Key: ASTERISK-25676
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-25676
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Affects Versions: 13.7.0
> Environment: Debian 8.2
> Reporter: Peter Sokolov
>
> The call originating SIP device only has g722 in the list of allowed codecs and directmedia=no set in its configuration. On a call to SIP destination that replies with alaw (183 and OK), Asterisk 13 will correctly transcode alaw to g722 during the early media, however as soon as the connection is established (OK) and the bridge switches to native_rtp, it will simply let through the alaw packets.
> To reproduce, configure like below. 333 is a SIP phone (FritzBox in my case) that supports g722 and alaw. OutTrunk is a SIP trunk provider that does not support g722 for calls out of its network (it replies with alaw only in 183 and OK SDP). At least one peer has to be set to directmedia=no in a way that RTP always flows via Asterisk.
> Make a call from peer 333 to a number that is routed through OutTrunk. When OutTrunk sends 183 with only alaw in SDP, you will see that Asterisk sends 183 with g722 in SDP to peer 333 and will transcode the alaw early media to g722. As soon as the connection is established and OutTrunk sends OK with only alaw in SDP, you will see that Asterisk sends OK with g722 in SDP to peer 333, however native_rtp bridge will let through the alaw media to the peer 333 even though alaw is not in the list of allowed codecs for the peer 333.
> {code}
> sip.conf:
> [333]
> type=friend
> context=users
> subscribecontext=users
> secret=donttell
> language=de
> mailbox=333 at default
> host=dynamic
> dtmfmode=rfc2833
> defaultuser=333
> callerid=333 <00333>
> restrictcid=no
> fromdomain=test.tst
> directmedia=no
> nat=no
> insecure=port
> call-limit=5
> allowsubscribe=yes
> disallow=all
> allow=g722
> [OutTrunk]
> type=peer
> fromdomain=test2.tst
> defaultuser=user1
> secret=donttell
> host=test2.tst
> context=users
> directmedia=yes
> qualify=no
> insecure=port,invite
> promiscredir=no
> dtmfmode=rfc2833
> nat=no
> disallow=all
> allow=g722,alaw
> callgroup=1
> t38pt_udptl=no
> t38pt_rtp=no
> t38pt_tcp=no
> extensions.conf:
> [users]
> exten => 123,1,NoOp
> exten => 123,n,Set(CALLERID(number)=00333)
> exten => 123,n,Set(CALLERID(name)=${CALLERID(number):2})
> exten => 123,n,Set(CALLERID(number)=${CALLERID(number):2})
> exten => 123,n,Dial(SIP/${EXTEN}@OutTrunk,60,)
> exten => 123,n,Hangup
> {code}
> Here is the log where "Sent RTP P2P packet" forwarding the alaw packet without transcoding can be seen:
> {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=z9hG4bK7D8CF25D90C364B9
> From: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
> To: <sip:123 at test.tst>
> Call-ID: 241FB38ED47A0E77 at 192.168.0.30
> CSeq: 2573 INVITE
> Contact: <sip:333 at X.X.X.X;uniq=C09040DC74C0F2A9D6039136CB478>
> Max-Forwards: 70
> Expires: 120
> User-Agent: AVM FRITZ!Box 7330 107.06.30 (Jul 10 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: 433
> v=0
> o=user 9081636 9081636 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 97 120 121 101
> 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:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:120 PCMA/16000
> a=rtpmap:121 PCMU/16000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> 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 - 241FB38ED47A0E77 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 97
> Found RTP audio format 120
> Found RTP audio format 121
> Found RTP audio format 101
> 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 iLBC for ID 97
> Found unknown media description format PCMA for ID 120
> Found unknown media description format PCMU for ID 121
> Found audio description format telephone-event for ID 101
> 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;uniq=C09040DC74C0F2A9D6039136CB478>
> <--- 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=z9hG4bK7D8CF25D90C364B9;received=X.X.X.X
> f: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
> t: <sip:123 at test.tst>
> i: 241FB38ED47A0E77 at 192.168.0.30
> CSeq: 2573 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/333-000001f3", "") in new stack
> -- Executing [123 at users:2] NoOp("SIP/333-000001f3", "") in new stack
> -- Executing [123 at users:3] Set("SIP/333-000001f3", "CALLERID(name)=333") in new stack
> -- Executing [123 at users:4] Set("SIP/333-000001f3", "CALLERID(number)=333") in new stack
> -- Executing [123 at users:5] Dial("SIP/333-000001f3", "SIP/123 at OutTrunk,60,") in new stack
> == Using SIP RTP CoS mark 5
> Audio is at 12608
> 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=z9hG4bK5b61c9e2
> Max-Forwards: 70
> f: "333" <sip:333 at test2.tst>;tag=as4c31698e
> t: <sip:123 at test2.tst>
> m: <sip:333 at A.A.A.A:5060>
> i: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
> CSeq: 102 INVITE
> User-Agent: Asterisk
> Date: Sat, 09 Jan 2016 21:27:20 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> k: replaces, timer
> c: application/sdp
> l: 279
> v=0
> o=root 1956784559 1956784559 IN IP4 A.A.A.A
> s=Asterisk PBX 13.7.0-rc2
> c=IN IP4 A.A.A.A
> t=0 0
> m=audio 12608 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=z9hG4bK5b61c9e2
> From: "333" <sip:333 at test2.tst>;tag=as4c31698e
> To: <sip:123 at test2.tst>
> Contact: sip:123 at D.D.D.D:5060
> Call-ID: 669e68ca24c091413a62b0e06a0c2128 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=z9hG4bK5b61c9e2
> From: "333" <sip:333 at test2.tst>;tag=as4c31698e
> To: <sip:123 at test2.tst>;tag=360313ac55ed93a8b99b71
> Contact: sip:123 at D.D.D.D:5060
> Call-ID: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
> CSeq: 102 INVITE
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Type: application/sdp
> Content-Length: 212
> v=0
> o=CARRIER 1452374900 1452374900 IN IP4 G.G.G.G
> s=SIP Call
> c=IN IP4 G.G.G.G
> t=0 0
> m=audio 26660 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:123 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 G.G.G.G:26660
> -- SIP/OutTrunk-000001f4 is making progress passing it to SIP/333-000001f3
> Audio is at 17508
> Adding codec g722 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=z9hG4bK7D8CF25D90C364B9;received=X.X.X.X
> f: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
> t: <sip:123 at test.tst>;tag=as521bc73b
> i: 241FB38ED47A0E77 at 192.168.0.30
> CSeq: 2573 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: 253
> v=0
> o=root 486862468 486862468 IN IP4 A.A.A.A
> s=Asterisk PBX 13.7.0-rc2
> c=IN IP4 A.A.A.A
> t=0 0
> m=audio 17508 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <------------>
> > 0x27e1720 -- Probation passed - setting RTP source address to G.G.G.G:26660
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026726, ts 3738337846, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007407, ts 000120, len 000160)
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026727, ts 3738338006, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007408, ts 000280, len 000160)
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026728, ts 3738338166, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007409, ts 000440, len 000160)
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026729, ts 3738338326, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007410, ts 000600, len 000160)
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026730, ts 3738338486, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007411, ts 000760, len 000160)
> > 0x253cd30 -- Probation passed - setting RTP source address to X.X.X.X:7078
> Got RTP packet from X.X.X.X:7078 (type 09, seq 000001, ts 000240, len 000160)
> Sent RTP packet to G.G.G.G:26660 (type 08, seq 049922, ts 000152, len 000160)
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026731, ts 3738338646, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007412, ts 000920, len 000160)
> Got RTP packet from X.X.X.X:7078 (type 09, seq 000002, ts 000480, len 000240)
> Sent RTP packet to G.G.G.G:26660 (type 08, seq 049923, ts 000392, len 000240)
> Got RTP packet from G.G.G.G:26660 (type 08, seq 026732, ts 3738338806, len 000160)
> Sent RTP packet to X.X.X.X:7078 (type 09, seq 007413, ts 001080, len 000160)
> <--- SIP read from UDP:D.D.D.D:5060 --->
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP 172.31.1.100:5060;branch=z9hG4bK5b61c9e2
> From: "333" <sip:333 at test2.tst>;tag=as4c31698e
> To: <sip:123 at test2.tst>;tag=360313ac55ed93a8b99b71
> Contact: sip:123 at D.D.D.D:5060
> Call-ID: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
> CSeq: 102 INVITE
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Type: application/sdp
> Content-Length: 212
> v=0
> o=CARRIER 1452374913 1452374913 IN IP4 G.G.G.G
> s=SIP Call
> c=IN IP4 G.G.G.G
> t=0 0
> m=audio 26660 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=ptime:20
> a=sendrecv
> <------------->
> --- (11 headers 10 lines) ---
> 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 G.G.G.G:26660
> sip_route_dump: route/path hop: <sip:123 at D.D.D.D:5060>
> set_destination: Parsing <sip:123 at D.D.D.D:5060> for address/port to send to
> set_destination: set destination to D.D.D.D:5060
> Transmitting (no NAT) to D.D.D.D:5060:
> ACK sip:123 at D.D.D.D:5060 SIP/2.0
> v: SIP/2.0/UDP A.A.A.A:5060;branch=z9hG4bK0442d057
> Max-Forwards: 70
> f: "333" <sip:333 at test2.tst>;tag=as4c31698e
> t: <sip:123 at test2.tst>;tag=360313ac55ed93a8b99b71
> m: <sip:333 at A.A.A.A:5060>
> i: 669e68ca24c091413a62b0e06a0c2128 at test2.tst
> CSeq: 102 ACK
> User-Agent: Asterisk
> l: 0
> ---
> -- SIP/OutTrunk-000001f4 answered SIP/333-000001f3
> Audio is at 17508
> Adding codec g722 to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Reliably Transmitting (no NAT) to X.X.X.X:5060 --->
> SIP/2.0 200 OK
> v: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bK7D8CF25D90C364B9;received=X.X.X.X
> f: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
> t: <sip:123 at test.tst>;tag=as521bc73b
> i: 241FB38ED47A0E77 at 192.168.0.30
> CSeq: 2573 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: 253
> v=0
> o=root 486862468 486862468 IN IP4 A.A.A.A
> s=Asterisk PBX 13.7.0-rc2
> c=IN IP4 A.A.A.A
> t=0 0
> m=audio 17508 RTP/AVP 9 101
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <------------>
> -- Channel SIP/OutTrunk-000001f4 joined 'simple_bridge' basic-bridge <94935aef-db90-4ff7-ac85-60d04663af6d>
> -- Channel SIP/333-000001f3 joined 'simple_bridge' basic-bridge <94935aef-db90-4ff7-ac85-60d04663af6d>
> > Bridge 94935aef-db90-4ff7-ac85-60d04663af6d: switching from simple_bridge technology to native_rtp
> > Locally RTP bridged 'SIP/333-000001f3' and 'SIP/OutTrunk-000001f4' in stack
> > Locally RTP bridged 'SIP/333-000001f3' and 'SIP/OutTrunk-000001f4' in stack
> > 0x27e1720 -- Probation passed - setting RTP source address to G.G.G.G:26660
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> <--- SIP read from UDP:X.X.X.X:5060 --->
> ACK sip:123 at 172.31.1.100:5060 SIP/2.0
> Via: SIP/2.0/UDP X.X.X.X:5060;branch=z9hG4bKF1AA2AEE2EEA0175
> From: <sip:333 at test.tst>;tag=A4D1A0E52D14A452
> To: <sip:123 at test.tst>;tag=as521bc73b
> Call-ID: 241FB38ED47A0E77 at 192.168.0.30
> CSeq: 2573 ACK
> Contact: <sip:333 at X.X.X.X;uniq=C09040DC74C0F2A9D6039136CB478>
> Max-Forwards: 70
> User-Agent: AVM FRITZ!Box 7330 107.06.30 (Jul 10 2015)
> Content-Length: 0
> <------------->
> --- (10 headers 0 lines) ---
> Got RTP packet from X.X.X.X:7078 (type 09, seq 000409, ts 098160, len 000240)
> Sent RTP packet to G.G.G.G:26660 (type 08, seq 050330, ts 098072, len 000240)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Got RTP packet from X.X.X.X:7078 (type 09, seq 000410, ts 098560, len 000400)
> Sent RTP packet to G.G.G.G:26660 (type 08, seq 050331, ts 098472, len 000400)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> Got RTP packet from X.X.X.X:7078 (type 09, seq 000411, ts 098800, len 000240)
> Sent RTP packet to G.G.G.G:26660 (type 08, seq 050332, ts 098712, len 000240)
> Sent RTP P2P packet to X.X.X.X:7078 (type 08, len 000160)
> {code}
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