[Asterisk-Users] Cisco to Cisco - poor quality

[Terence Parker]

I have managed to find time to have another go at the Cisco phones - 
alas, I am still having problems with Cisco to Cisco calls.

Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have 
tried setting both phones to different codecs (tried default g729a, 
g711alaw, and g711ulaw). Also, the other observations that have been 
made:

- Problem is one-way. One side hears me clearly ; I don't hear the 
other side clearly at all (5% audible only).
- Calls to MSN are fine (two way conversation is crystal clear)
- Calls to a Zultys Zip2 SIP phone is also perfectly clear.
- All these three tested over the same network and same VPN (call 
between Hong Kong and USA).
- Cisco to Cisco calls worked fine with Vocal.

If Cisco is able to talk fine with other devices, there should not be a 
problem with bandwidth or my network. However, I am finding it quite 
bizzarre that Cisco is unable to talk to itself. The problem shouldn't 
be VAD or the like - even if I talk non-stop, or the other guy does, I 
get the same problem.

I attach a copy of my Cisco phone configuration for reference. I have 
even recently upgraded my phone firmware - but no luck.

Platform : Cisco IP Phone 7960
Elasped Time: 08:11:26

dhcp_server : 192.168.8.254
my_ip_addr : 192.168.8.83
subnet_mask : 255.255.255.0
defaultgw : 192.168.8.254
dyn_dns_addr_1 : 0.0.0.0
dyn_dns_addr_2 : 0.0.0.0
dns_addr : 205.252.144.228
dns_backup_1: 202.14.67.4
tftp_addr : 192.168.0.252
dyn_tftp_addr : 0.0.0.0
my_mac_addr : 0007:50ac:6932
domain_name : deltapath.com
my_name : SIP000750AC6932
Status Flags : 12300000

image_version : "P0S3-05-3-00"
FirmLoadID : "PC03A300"
network_media_type : Auto
network_port2_type : Hub/Switch
tos_media : 5
phone_label : "DELTAPATH"
tftp_cfg_dir : "./sip_phone/"
phone_password : **********
phone_prompt : "SIP Phone"
language : english
sntp_mode : DirectedBroadcast
sntp_server : stdtime.gov.hk
time_zone : HST
dst_offset : 0
dst_start_month : April
dst_start_day : 0
dst_start_day_of_week : Sun
dst_start_week_of_month : 1
dst_start_time : 02
dst_stop_month : Oct
dst_stop_day : 0
dst_stop_day_of_week : Sunday
dst_stop_week_of_month : 8
dst_stop_time : 2
dst_auto_adjust : 0
time_format_24hr : 1
date_format : M/D/Y
nat_enable : 0
nat_address :
voip_control_port : 5060
start_media_port : 16384
end_media_port : 32766
sync : "1"
xml_card_dir : ""
xml_card_file : "CARD.XML"
telnet_level : 2
services_url : ""
directory_url : ""
logo_url : "http://deltapath.com/logo.bmp"
http_proxy_addr :
http_proxy_port : 80
enable_vad : 0
dial_template : "dialplan"
callerid_blocking : 0
anonymous_call_block : 0
autocomplete : 1
messages_uri : "86"
dnd_control : 0
preferred_codec : g729a
dtmf_outofband : avt
dtmf_avt_payload : 101
dtmf_db_level : 3
dtmf_inband : 1
line1_name : "TerenceParker"
line2_name : "74xxx"
line3_name : "74xxx"
line4_name : ""
line5_name : ""
line6_name : ""
line1_authname : "TerenceParker"
line2_authname : "74xxx"
line3_authname : "74xxx"
line4_authname : "UNPROVISIONED"
line5_authname : "UNPROVISIONED"
line6_authname : "UNPROVISIONED"
line1_shortname : "Asterisk"
line2_shortname : "FWD-74xxx"
line3_shortname : "FWD-74xxx"
line4_shortname : "UNPROVISIONED"
line5_shortname : "UNPROVISIONED"
line6_shortname : "UNPROVISIONED"
line1_displayname : "TerenceParker"
line2_displayname : "74xxx"
line3_displayname : "Terence Parker"
line4_displayname : ""
line5_displayname : ""
line6_displayname : ""
proxy1_address : "192.168.0.254"
proxy2_address : "fwd.pulver.com"
proxy3_address : "fwd.pulver.com"
proxy4_address : ""
proxy5_address : ""
proxy6_address : ""
proxy1_port : 5060
proxy2_port : 5060
........
sip_retx : 10
sip_invite_retx : 6
timer_t1 : 500
timer_t2 : 4000
timer_invite_expires : 180
timer_register_expires : 3600
proxy_register : 1
proxy_backup : "UNPROVISIONED"
proxy_emergency : "UNPROVISIONED"
proxy_backup_port : 0
proxy_emergency_port : 0
outbound_proxy :
outbound_proxy_port : 5082
nat_received_processing : 0
mwi_status : 0
call_waiting : 1
user_info : none
cnf_join_enable : 1
remote_party_id : 0
semi_attended_transfer : 1
call_hold_ringback : 0


Thanks for any help!

Terence


> I have never used Cisco phones, but I have had problems in the past
> relating to * RTP talking to a widget with VAD turned on.
> * RTP stack can not run on its own.  It relies on receiving RTP packets
> for doing its timing.
>
> A simple test is to sniff the line to make sure the phones always send 
> packets.
> If you see pauses, you may need to disable some type of VAD setting on 
> the phone.
> Or just never quit talking when using the Cisco phone.
>
> Terence Parker wrote:
>
>> I have set canreinvite=no in the sip.conf for each user (well, there 
>> are
>> only two) using a cisco phone. What does this imply?
>>
>> As for whether the problem is due to the phones or asterisk however,
>> indications would suggest both, because:
>>
>> - Voicemail works fine (and is clear)
>> - I can initiate a call between MSN and Cisco, and that would sound 
>> fine.
>>
>> This might suggest a problem with my phones. However :
>>
>>    -  When using Vocal previously, Cisco to Cisco conversation was 
>> fine.
>>
>> This has led me to be completely stumped! I notice some mention 
>> elsewhere
>> about asterisk lacking certain codecs because of license 
>> restrictions? Is
>> this anything to do with me? Or should the phones still - in theory - 
>> be
>> able to talk to each other without any problems? I have tried the 
>> cisco
>> phone on both g729a and g711ulaw.
>>
>> I'm currently *trying* to get ahold of an updated firmware for my 
>> phone. I
>> will see if this fixes the problems.
>>
>> Thanks again,
>>
>> Terence