a.ahmad@dsl.pipex.com
2004-Jul-13 10:36 UTC
[Asterisk-Users] codec issues between linphone and *
Hello
I am trying to connect linphone 0.12.2 to an * 0.9.1 box over a LAN using the
console version of linphone. both boxs are using the latest alsa drivers on a
LFS kernal 2.4. I am running into errors with codec compatability between
linphone and *.
A point to note is that I am able to connect to asterisk using other sip
phones noteably sjphone however linephone is giving me some problems.
I can even connect from the linphone console to asterisk and hear the default
message however the connection dies on linphones side but the default message
completes to the end. I get the following output:
| INFO1 | <ict_callbacks.c: 30> Transaction 1 killed.
I have have configured the .linphone config file (as shown below) to use
various codecs pcmu gsm & speex and in sip.conf on * to use specific codes
to
no avail.
I have some Q which are still unclear to me despite sifting through the
archives.
Q1 what is the dtmfmode based on ( i read that it is determined according to
the phone being use and i have also read that it is base on he codec being
used, rfc2833 for alaw/ulaw info/outband for others. Does this need to be set
for me to carry out a test from linphonec to * default message?
Q2 the rtp ports used by asterisk rang from 10000/20000 while linphone uses
7078, which I changed to 17078 to allow it to fall within the range. however I
notice during the invite/ok trans. that rtp uses 18XXX and then uses 0 when no
codec can be agreed upon.
Q3 Does sip and rtp negotiate codes independently of eachother I have noticed
and read the rtp seems to have a choice of pcmu/gsm/speex despite me having
asked it to use them. can these be set/controlled from somewhere.
I have yet to find a webpage explaining the possible setting of a linphonec
config file. I am not sure as to it is using ~/.linphonec or ~/.gnome2/
linphone config file.
I have also seen that there is a sdp patch for when linphone doesn't accept
a
codec (e.g. a payload type), the SDP frame respond with a port equal to 0, but
the payload type correspondingis not sent. would this help me with my
problems?
patch link: http://lists.gnu.org/archive/html/linphone-users/2004-07/
msg00009.html
I have attached the the output from linephonec debug output, .linphonec config
and my asterisk sip.conf file below. any sort of advice would be appreciated
thanks in advance. Amjad
how would i reply to an answer if I have to? do i email replies or is it
conmtrolled from within the thread? thanks
*****************asterisk output*******************
CLI> reloaD
Jul 13 18:25:14 WARNING[16384]: chan_iax2.c:5689 set_config: Ignoring port for
now
Jul 13 18:25:14 NOTICE[16384]: indications.c:394
ast_unregister_indication_country: Removed default indication country
'uk'
*CLI>
Sip read:
INVITE sip:1000@192.168.10.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK2213492093
From: <sip:aa@192.168.10.24>;tag=2982198999
To: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 20 INVITE
Contact: <sip:aa@192.168.10.24>
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Type: application/sdp
Content-Length: 212
v=0
o=aa 123456 654321 IN IP4 192.168.10.24
s=A conversation
c=IN IP4 192.168.10.24
t=0 0
m=audio 7078 RTP/AVP 110 101
b=AS:8
a=rtpmap:110 speex/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
11 headers, 10 lines
Using latest request as basis request
Sending to 192.168.10.24 : 5060 (non-NAT)
Found audio format UNKN
Found audio format UNKN
Found description format speex
Found description format telephone-event
Capabilities: us - 524814, them - 512/0, combined - 512
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 1000 in default
list_route: hop: <sip:aa@192.168.10.24>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK2213492093
From: <sip:aa@192.168.10.24>;tag=2982198999
To: <sip:1000@192.168.10.20>;tag=as062515d4
Call-ID: 728714378@192.168.10.24
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1000@192.168.10.20>
Content-Length: 0
to 192.168.10.24:5060
We're at 192.168.10.20 port 17090
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with capability 512
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK2213492093
From: <sip:aa@192.168.10.24>;tag=2982198999
To: <sip:1000@192.168.10.20>;tag=as062515d4
Call-ID: 728714378@192.168.10.24
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1000@192.168.10.20>
Content-Type: application/sdp
Content-Length: 236
v=0
o=root 4112 4112 IN IP4 192.168.10.20
s=session
c=IN IP4 192.168.10.20
t=0 0
m=audio 17090 RTP/AVP 3 0 8 110
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 SPEEX/8000
a=silenceSupp:off - - - -
to 192.168.10.24:5060
Sip read:
ACK sip:1000@192.168.10.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.24:5060;branch=z9hG4bK2088217008
From: <sip:aa@192.168.10.24>;tag=2982198999
To: <sip:1000@192.168.10.20>;tag=as062515d4
Call-ID: 728714378@192.168.10.24
CSeq: 20 ACK
max-forwards: 10
user-agent: oSIP/Linphone-0.12.1
Content-Length: 0
9 headers, 0 lines
Jul 13 18:25:38 NOTICE[311311]: channel.c:1508 ast_set_read_format: Unable to
find a path from SPEEX to ULAW
Jul 13 18:25:38 NOTICE[311311]: channel.c:1478 ast_set_write_format: Unable to
find a path from GSM to SPEEX
Jul 13 18:25:38 WARNING[311311]: chan_sip.c:1333 sip_write: Asked to transmit
frame type 4, while native formats is 512 (read/write = 4/2)
Jul 13 18:25:38 WARNING[311311]: file.c:538 ast_readaudio_callback: Failed to
write frame
Jul 13 18:25:38 NOTICE[311311]: channel.c:1478 ast_set_write_format: Unable to
find a path from ULAW to SPEEX
Jul 13 18:25:38 WARNING[311311]: file.c:171 ast_stopstream: Unable to restore
format back to 4
set_destination: Parsing <sip:aa@192.168.10.24> for address/port to send
to
set_destination: set destination to 192.168.10.24, port 5060
Reliably Transmitting:
BYE sip:aa@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.20:5060;branch=z9hG4bK1b7c2ff1
From: <sip:1000@192.168.10.20>;tag=as062515d4
To: <sip:aa@192.168.10.24>;tag=2982198999
Contact: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 192.168.10.24:5060
Retransmitting #1 (no NAT):
BYE sip:aa@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.20:5060;branch=z9hG4bK1b7c2ff1
From: <sip:1000@192.168.10.20>;tag=as062515d4
To: <sip:aa@192.168.10.24>;tag=2982198999
Contact: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.168.10.24:5060
Retransmitting #2 (no NAT):
BYE sip:aa@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.20:5060;branch=z9hG4bK1b7c2ff1
From: <sip:1000@192.168.10.20>;tag=as062515d4
To: <sip:aa@192.168.10.24>;tag=2982198999
Contact: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.168.10.24:5060
Retransmitting #3 (no NAT):
BYE sip:aa@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.20:5060;branch=z9hG4bK1b7c2ff1
From: <sip:1000@192.168.10.20>;tag=as062515d4
To: <sip:aa@192.168.10.24>;tag=2982198999
Contact: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.168.10.24:5060
Retransmitting #4 (no NAT):
BYE sip:aa@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.20:5060;branch=z9hG4bK1b7c2ff1
From: <sip:1000@192.168.10.20>;tag=as062515d4
To: <sip:aa@192.168.10.24>;tag=2982198999
Contact: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.168.10.24:5060
Retransmitting #5 (no NAT):
BYE sip:aa@192.168.10.24 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.20:5060;branch=z9hG4bK1b7c2ff1
From: <sip:1000@192.168.10.20>;tag=as062515d4
To: <sip:aa@192.168.10.24>;tag=2982198999
Contact: <sip:1000@192.168.10.20>
Call-ID: 728714378@192.168.10.24
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
to 192.168.10.24:5060
Jul 13 18:25:44 WARNING[213006]: chan_sip.c:497 retrans_pkt: Maximum retries
exceeded on call 728714378@192.168.10.24 for seqno 102 (Request)
*CLI>
*****************sip.conf*************************
[general]
port=5060 ; Port to bind to
bindaddr=192.168.10.20 ; Address to bind SIP channel to
context=default ; Default context for incoming calls
;localnet=192.168.10.0 ; Internet NETWORK address
;localmask=255.255.255.0 ; Internet netmask
;srvlookup = yes ; Enable DNS SRV lookups on outbound calls
;pedantic = yes ; Enable slow, pedantic checking for Pingtel
;tos=lowdelay ; IP QoS parameter, either keyword or value
;maxexpirey=3600 ; Max length of incoming registration we allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY
;videosupport=yes ; Turn on support for SIP video
dissallow=all ; Disallow all codecs
;allow=all
;allow=speex
;allow=ulaw
;allow=gsm ; Allow codecs in order of preference
:allow=ilbc
*****************.linphonec config file ****************************
[net]
if_name=rhine
con_type=1
use_nat=0
[sip]
username=aa
hostname=192.168.10.24
sip_port=5060
use_registrar=0
as_proxy=0
expires=900
[sound]
dev_id=1
rec_lev=80
play_lev=80
source=m
local_ring=/usr/share/sounds/linphone/rings/oldphone.wav
remote_ring=/usr/share/sounds/linphone/ringback.wav
[rtp]
audio_rtp_port=17078
video_rtp_port=0
audio_jitt_comp=60
video_jitt_comp=60
[video]
enabled=0
show_local=0
[audio_codec_0]
mime=PCMU
rate=8000
enabled=0
[audio_codec_3]
mime=GSM
rate=8000
enabled=0
[audio_codec_2]
mime=PCMA
rate=8000
enabled=0
[audio_codec_1]
mime=speex
rate=8000
enabled=1
[audio_codec_4]
mime=speex
rate=16000
enabled=0
[audio_codec_5]
mime=1015
rate=8000
enabled=0
[address_book]
entry_count=0
--
Harold Workman
2004-Jul-13 13:24 UTC
[Asterisk-Users] Problem with multiple phones behind firewall
Hi, I am having a problem when I add multiple phones behind a Symmetric Firewall. Heres my situation. 11am - Phone A registers with * 11:01am - test call to Phone A. Call works fine. 11:02am - Phone B registers with * 11:03am - test call to Phone A fails, test call to phone B works fine. 11:04am - test call from Phone A to Phone B and vice versa works fine. 11:05am - Phone A re-registers with *. Test call to Phone A works fine now. This happens on almost all occasions. When I see one phone register behind a firewall, i then see the "Retransmitting #5 (NAT):" messages, until I received the "Jul 13 15:11:09 WARNING[1133718080]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 0576f80274aebde8796e5d4b2444c2a9@64.72.107.10 for seqno 102 (Non-critical Request)" I have nat=yes in my sip.conf file. I have tried using the qualify command, but I have never been able to get it to work behind a symmetric firewall to both a unknown sip phone and xlite. The moment I turn on qualify, I see the Options request sent out, and on the client see the options request, but I never see a response on * from the clients. Here is what my sip.conf looks like... [general] port = 5060 bindaddr = 64.72.107.10 context = exten maxexpirey=3000 defaultexpirey=300 disallow=all allow=alaw allow=ulaw [123456] type=friend secret=k3v1n nat=yes canreinvite=no host=dynamic dtmfmode=rfc2833 context=cytelmain [789012] type=friend secret=cytel nat=yes canreinvite=no host=dynamic dtmfmode=rfc2833 context=cytelmain What else is there for me to try to resolve my NAT problem with multiple users behind a symmetric firewall? Thanks, Harold