[asterisk-users] Codec Negotiation problem
research at businesstz.com
research at businesstz.com
Thu Jun 13 12:04:11 CDT 2013
Hi there
I have asterisk 10.11.1 which seems to have problem negotiating codec.
Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p
and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw,
h263p. I have tried similar combination of codecs and SIP phone but when
making a video call, it report "Peer doesn't provide video". It seems
Asterisk is failing to set capability correct. Both codecs are enabled on
the SIP Phones
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|h263p), 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 10.10.10.129:53188
Peer doesn't provide video
Here is a sip show peer output and log when making calls.
localhost*CLI> sip show peer 1003
* Name : 1003
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : video-users
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 1003 at device
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "device" <1003>
MaxCallBR : 384 kbps
Expire : 3605
Insecure : no
Force rport : Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 10.10.10.129:48464
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1003
SIP Options : (none)
Codecs : (alaw|h263p)
Codec Order : (alaw:20,h263p:0)
Auto-Framing : No
Status : OK (8 ms)
Useragent : X-Lite release 4.5.2 stamp 70142
Reg. Contact : sip:1003 at 10.10.10.129:48464;rinstance=cf0c3558f05c89dc
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
localhost*CLI> sip show peer 1004
* Name : 1004
Description :
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : video-users
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 1004 at device
VM Extension : asterisk
LastMsgsSent : 0/0
Call limit : 2147483647
Max forwards : 0
Dynamic : Yes
Callerid : "device" <1004>
MaxCallBR : 384 kbps
Expire : 893
Insecure : no
Force rport : Yes
ACL : Yes
DirectMedACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 10.10.10.107:21769
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 1004
SIP Options : (none)
Codecs : (alaw|h263p)
Codec Order : (alaw:20,h263p:0)
Auto-Framing : No
Status : OK (2 ms)
Useragent : Grandstream GXV3175v2 1.0.1.19
Reg. Contact : sip:1004 at 10.10.10.107:21769
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No
localhost*CLI>
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.10.10.129:48464 --->
INVITE sip:1004 at 10.10.10.105 SIP/2.0
Via: SIP/2.0/UDP
10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003 at 10.10.10.129:48464>
To: <sip:1004 at 10.10.10.105>
From: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.2 stamp 70142
Authorization: Digest
username="1003",realm="10.10.10.105",nonce="05e8af6e",uri="sip:1004 at 10.10.10.105",response="20e63a04aa86d6ec1d1e045c05159b39",algorithm=MD5
Content-Length: 418
v=0
o=- 13015615910543193 1 IN IP4 10.10.10.129
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 10.10.10.129
t=0 0
m=audio 53188 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 49490 RTP/AVP 115 34
a=rtpmap:115 H263-1998/90000
a=fmtp:115 QCIF=2;CIF=2;VGA=2;CIF4=2;I=1;J=1;T=1
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=2;CIF=2;VGA=2;CIF4=2
a=rtcp-fb:* nack pli
a=sendrecv
<------------->
--- (14 headers 16 lines) ---
Sending to 10.10.10.129:48464 (NAT)
Using INVITE request as basis request -
MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
Found peer '1003' for '1003' from 10.10.10.129:48464
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 115
Found RTP video format 34
Found video description format H263-1998 for ID 115
Found video description format H263 for ID 34
Capabilities: us - (alaw|h263p), peer -
audio=(ulaw|alaw)/video=(h263|h263p)/text=(nothing), combined -
(alaw|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.10.10.129:53188
Peer video RTP is at port 10.10.10.129:49490
Looking for 1004 in video-users (domain 10.10.10.105)
list_route: hop: <sip:1003 at 10.10.10.129:48464>
<--- Transmitting (NAT) to 10.10.10.129:48464 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;received=10.10.10.129;rport=48464
From: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
To: <sip:1004 at 10.10.10.105>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 2 INVITE
Server: Asterisk PBX 10.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:1004 at 10.10.10.105:5060>
Content-Length: 0
<------------>
-- Executing [1004 at video-users:1] Answer("SIP/1003-00000020", "") in
new stack
Audio is at 13410
Video is at 10.10.10.105:13834
Adding codec 100004 (alaw) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 10.10.10.129:48464 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.10.10.129:48464;branch=z9hG4bK-d8754z-25f65c322686d22e-1---d8754z-;received=10.10.10.129;rport=48464
From: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
To: <sip:1004 at 10.10.10.105>;tag=as24914503
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 2 INVITE
Server: Asterisk PBX 10.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:1004 at 10.10.10.105:5060>
Content-Type: application/sdp
Content-Length: 315
v=0
o=root 1557854096 1557854096 IN IP4 10.10.10.105
s=Asterisk PBX 10.11.1
c=IN IP4 10.10.10.105
b=CT:384
t=0 0
m=audio 13410 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 13834 RTP/AVP 115
a=rtpmap:115 h263-1998/90000
a=sendrecv
<------------>
<--- SIP read from UDP:10.10.10.129:48464 --->
ACK sip:1004 at 10.10.10.105:5060 SIP/2.0
Via: SIP/2.0/UDP
10.10.10.129:48464;branch=z9hG4bK-d8754z-2ee063296946cc3e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:1003 at 10.10.10.129:48464>
To: <sip:1004 at 10.10.10.105>;tag=as24914503
From: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 2 ACK
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- Executing [1004 at video-users:2] Dial("SIP/1003-00000020",
"SIP/1004") in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Called SIP/1004
-- SIP/1004-00000021 is ringing
-- SIP/1004-00000021 answered SIP/1003-00000020
-- Remotely bridging SIP/1003-00000020 and SIP/1004-00000021
set_destination: Parsing <sip:1003 at 10.10.10.129:48464> for address/port to
send to
set_destination: set destination to 10.10.10.129:48464
Audio is at 13410
Video is at 10.10.10.107:57822
Adding codec 100004 (alaw) to SDP
Adding video codec 200003 (h263p) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.10.129:48464:
INVITE sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 315
v=0
o=root 1557854096 1557854097 IN IP4 10.10.10.107
s=Asterisk PBX 10.11.1
c=IN IP4 10.10.10.107
b=CT:384
t=0 0
m=audio 53104 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 57822 RTP/AVP 115
a=rtpmap:115 h263-1998/90000
a=sendrecv
---
<--- SIP read from UDP:10.10.10.129:48464 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060
Contact: <sip:1003 at 10.10.10.129:48464>
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
From: <sip:1004 at 10.10.10.105>;tag=as24914503
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 234
v=0
o=- 13015615910543193 2 IN IP4 10.10.10.129
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 10.10.10.129
t=0 0
m=audio 53188 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 0 RTP/AVP 115
<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|h263p), 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 10.10.10.129:53188
Peer doesn't provide video
set_destination: Parsing <sip:1003 at 10.10.10.129:48464> for address/port to
send to
set_destination: set destination to 10.10.10.129:48464
Transmitting (NAT) to 10.10.10.129:48464:
ACK sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK24550383;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.11.1
Content-Length: 0
---
set_destination: Parsing <sip:1003 at 10.10.10.129:48464> for address/port to
send to
set_destination: set destination to 10.10.10.129:48464
Audio is at 13410
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.10.129:48464:
INVITE sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK23eb7489;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 1557854096 1557854098 IN IP4 10.10.10.107
s=Asterisk PBX 10.11.1
c=IN IP4 10.10.10.107
t=0 0
m=audio 53104 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
Retransmitting #1 (NAT) to 10.10.10.129:48464:
INVITE sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK23eb7489;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 1557854096 1557854098 IN IP4 10.10.10.107
s=Asterisk PBX 10.11.1
c=IN IP4 10.10.10.107
t=0 0
m=audio 53104 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:10.10.10.129:48464 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK23eb7489;rport=5060
Contact: <sip:1003 at 10.10.10.129:48464>
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
From: <sip:1004 at 10.10.10.105>;tag=as24914503
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 211
v=0
o=- 13015615910543193 3 IN IP4 10.10.10.129
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 10.10.10.129
t=0 0
m=audio 53188 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|h263p), 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 10.10.10.129:53188
Peer doesn't provide video
set_destination: Parsing <sip:1003 at 10.10.10.129:48464> for address/port to
send to
set_destination: set destination to 10.10.10.129:48464
Transmitting (NAT) to 10.10.10.129:48464:
ACK sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK2f83ceba;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.11.1
Content-Length: 0
---
set_destination: Parsing <sip:1003 at 10.10.10.129:48464> for address/port to
send to
set_destination: set destination to 10.10.10.129:48464
Audio is at 13410
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.10.129:48464:
INVITE sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK062304c6;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 104 INVITE
User-Agent: Asterisk PBX 10.11.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 1557854096 1557854099 IN IP4 10.10.10.107
s=Asterisk PBX 10.11.1
c=IN IP4 10.10.10.107
t=0 0
m=audio 53104 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:10.10.10.129:48464 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK062304c6;rport=5060
Contact: <sip:1003 at 10.10.10.129:48464>
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
From: <sip:1004 at 10.10.10.105>;tag=as24914503
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 104 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces, eventlist
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 211
v=0
o=- 13015615910543193 4 IN IP4 10.10.10.129
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 10.10.10.129
t=0 0
m=audio 53188 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|h263p), 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 10.10.10.129:53188
Peer doesn't provide video
set_destination: Parsing <sip:1003 at 10.10.10.129:48464> for address/port to
send to
set_destination: set destination to 10.10.10.129:48464
Transmitting (NAT) to 10.10.10.129:48464:
ACK sip:1003 at 10.10.10.129:48464 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK490ee170;rport
Max-Forwards: 70
From: <sip:1004 at 10.10.10.105>;tag=as24914503
To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
Contact: <sip:1004 at 10.10.10.105:5060>
Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA
CSeq: 104 ACK
User-Agent: Asterisk PBX 10.11.1
Content-Length: 0
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