Tek-Tips is the largest IT community on the Internet today!

Members share and learn making Tek-Tips Forums the best source of peer-reviewed technical information on the Internet!

  • Congratulations Chriss Miller on being selected by the Tek-Tips community for having the most helpful posts in the forums last week. Way to Go!

SIP early media 183 (SDP)

Status
Not open for further replies.

SAToronto

IS-IT--Management
Joined
Sep 21, 2011
Messages
199
Location
CA
I am having an issue with some sip trunks. The vendor seems to feel its an issue with the pbx
I have 5 ip 500s all config'd identical. Same ARS...same shortcodes same trunk config.
On this particular PBX I can make a long distance call via my sip trunks with out any issues. When we try to make a local or a toll free call, the call goes out...shows as connected, but I cant hear them and they cant here me. The vendors replied with::: The toll free call has early media 183(SDP) but it looks like they don’t know what to do with it. Ask if they can support early media?. My opinion I think would be that if I can make a good long distance call on the trunks I should be able to make a local or toll free also. they are trying to say its an issue with my ARS. My ARS doesnt care what type of call it is..all calls go out the same way.

Does anyone have any ideas about this or ever had an issue like this??

Thanks

Wayne
 
I am not sure but could this help?

· Codec Lockdown: Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP answer that also lists multiple codecs. This means that the user agent may switch to any of those codecs during the session without further negotiation. The system does not support multiple concurrent codecs for a session, so loss of speech path will occur if the codec is changed during the session. If codec lockdown is enabled, when the system receives an SDP answer with more than one codec from the list of offered codecs, it sends an extra re-INVITE using just a single codec from the list and resubmits a new SDP offer with just the single chosen codec.


BAZINGA!

I'm not insane, my mother had me tested!

 
Here is what the vendor sent for tracing


Toll Free no audio
Summary:
Request: Invite
Request URI: sip:1113218888402822@10.254.11.36
From: "name" <sip:1344@10.254.11.36>;tag=32c03a667e516938
To: <sip:1113218888402822@10.254.11.36>
Call-ID: e06a94d520b826932e9fe03fc843b83d
CSeq: 1723872074 INVITE
Contact: "name" <sip:1344@10.10.100.10:5060;transport=udp>
P-Asserted-Identity: "S+A Eng." <sip:7783720103@10.10.100.10:5060>
SDP IP: 10.10.100.10
SDP Port: 49154

Detail Data:

---- ETH ----
Destination Address: 00-1B-21-02-CB-DC
Source Address: 00-12-01-A1-D2-00
---- 802.1Q Virtual LAN ----
81-00 = 8100h [33024d] Type
{100. ....}= 04h [004d] Priority
{...0 ....}= 00h [000d] CFI
ID:
{.... 0001}= 01h [001d] bits 11-8
{1111 0100}= F4h [244d] bits 7-0
ID = 1F4h [500d]
08-00 = 800h [2048d] Protocol Type: Internet IP (IPv4)
---- IPv4 ----
---- IP header ----
{0100 ....}= 04h [004d] Version: IP Internet Protocol
{.... 0101}= 05h [005d] Header length: in 32 bit units - must be 5 or more
---- Type of Service ----
{100. ....}= 04h [004d] Precedence: Flash override
{...0 ....}= 00h [000d] Delay: Normal
{.... 1...}= 01h [001d] Throughput: High
{.... .0..}= 00h [000d] Reliability: Normal
{.... ..0.}= 00h [000d] ECN Capable Transport(ECT): No
{.... ...0}= 00h [000d] Congestion Experienced(CE): No
03-CF = 3CFh [975d] Total length: in octets, including header length - must not be less than 20
74-AE = 74AEh [29870d] Identification
{0... ....}= 00h [000d] reserved: valid
{.0.. ....}= 00h [000d] Don't Fragment (DF bit): May Fragment
{..0. ....}= 00h [000d] More Fragments (MF bit)
Fragment offset:
{...0 0000}= 00h [000d] bits 12-8
{0000 0000}= 00h [000d] bits 7-0
Fragment offset = 00h [000d]: in 64 bits units
{0101 1111}= 5Fh [095d] Time to live: Hops
{0001 0001}= 11h [017d] protocol: UDP (User Datagram)
5E-B2 = 5EB2h [24242d] Header checksum: checksum is correct
source ip address: 10.10.100.10
destination ip address: 10.254.11.36
---- IP datagram ----
---- UDP ----
---- UDP header ----
13-C4 = 13C4h [5060d] source port
13-C4 = 13C4h [5060d] destination port
03-BB = 3BBh [955d] length: octets include header (not less than 8)
00-00 = 00h [000d] checksum: checksum is not present
---- UDP datagram ----
---- SIP ----
INVITE sip:1113218888402822@10.254.11.36 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.10:5060;rport;branch=z9hG4bKb574c56f6aa81020d7d5eba8fab412ec
From: "name" <sip:1344@10.254.11.36>;tag=32c03a667e516938
To: <sip:1113218888402822@10.254.11.36>
Call-ID: e06a94d520b826932e9fe03fc843b83d
CSeq: 1723872074 INVITE
Contact: "name" <sip:1344@10.10.100.10:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer,100rel
User-Agent: IP Office 8.1 (56)
P-Asserted-Identity: "S+A Eng." <sip:7783720103@10.10.100.10:5060>
Content-Length: 298

v=0
o=UserA 2700914256 771711307 IN IP4 10.10.100.10
s=Session SDP
c=IN IP4 10.10.100.10
t=0 0
m=audio 49154 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Local no audio
Summary:
Request: Invite
Request URI: sip:111327807010331@10.254.11.36
From: "name" <sip:1344@10.254.11.36>;tag=0a2e9998d54978d1
To: <sip:111327807010331@10.254.11.36>
Call-ID: 5170b22c7de659a4dece187dbc4249c9
CSeq: 2006909703 INVITE
Contact: "name" <sip:1344@10.10.100.10:5060;transport=udp>
P-Asserted-Identity: "S+A Eng." <sip:7783720103@10.10.100.10:5060>
SDP IP: 10.10.100.10
SDP Port: 49154

Detail Data:

---- ETH ----
Destination Address: 00-1B-21-02-CB-DC
Source Address: 00-12-01-A1-D2-00
---- 802.1Q Virtual LAN ----
81-00 = 8100h [33024d] Type
{100. ....}= 04h [004d] Priority
{...0 ....}= 00h [000d] CFI
ID:
{.... 0001}= 01h [001d] bits 11-8
{1111 0100}= F4h [244d] bits 7-0
ID = 1F4h [500d]
08-00 = 800h [2048d] Protocol Type: Internet IP (IPv4)
---- IPv4 ----
---- IP header ----
{0100 ....}= 04h [004d] Version: IP Internet Protocol
{.... 0101}= 05h [005d] Header length: in 32 bit units - must be 5 or more
---- Type of Service ----
{100. ....}= 04h [004d] Precedence: Flash override
{...0 ....}= 00h [000d] Delay: Normal
{.... 1...}= 01h [001d] Throughput: High
{.... .0..}= 00h [000d] Reliability: Normal
{.... ..0.}= 00h [000d] ECN Capable Transport(ECT): No
{.... ...0}= 00h [000d] Congestion Experienced(CE): No
03-CE = 3CEh [974d] Total length: in octets, including header length - must not be less than 20
7F-86 = 7F86h [32646d] Identification
{0... ....}= 00h [000d] reserved: valid
{.0.. ....}= 00h [000d] Don't Fragment (DF bit): May Fragment
{..0. ....}= 00h [000d] More Fragments (MF bit)
Fragment offset:
{...0 0000}= 00h [000d] bits 12-8
{0000 0000}= 00h [000d] bits 7-0
Fragment offset = 00h [000d]: in 64 bits units
{0101 1111}= 5Fh [095d] Time to live: Hops
{0001 0001}= 11h [017d] protocol: UDP (User Datagram)
53-DB = 53DBh [21467d] Header checksum: checksum is correct
source ip address: 10.10.100.10
destination ip address: 10.254.11.36
---- IP datagram ----
---- UDP ----
---- UDP header ----
13-C4 = 13C4h [5060d] source port
13-C4 = 13C4h [5060d] destination port
03-BA = 3BAh [954d] length: octets include header (not less than 8)
00-00 = 00h [000d] checksum: checksum is not present
---- UDP datagram ----
---- SIP ----
INVITE sip:111327807010331@10.254.11.36 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.10:5060;rport;branch=z9hG4bK256dbde15678b0aba2e8d46dd4245fc2
From: "name" <sip:1344@10.254.11.36>;tag=0a2e9998d54978d1
To: <sip:111327807010331@10.254.11.36>
Call-ID: 5170b22c7de659a4dece187dbc4249c9
CSeq: 2006909703 INVITE
Contact: "name" <sip:1344@10.10.100.10:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer,100rel
User-Agent: IP Office 8.1 (56)
P-Asserted-Identity: "S+A Eng." <sip:7783720103@10.10.100.10:5060>
Content-Length: 299

v=0
o=UserA 3952385945 1413827740 IN IP4 10.10.100.10
s=Session SDP
c=IN IP4 10.10.100.10
t=0 0
m=audio 49154 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

LD good call
Summary:
Request: Invite
Request URI: sip:1113214164878151@10.254.11.36
From: "name" <sip:1344@10.254.11.36>;tag=36d16fdc706f3c4e
To: <sip:1113214164878151@10.254.11.36>
Call-ID: be3e816d5b84db55eac9cd62e22a83ff
CSeq: 989685916 INVITE
Contact: "name" <sip:1344@10.10.100.10:5060;transport=udp>
P-Asserted-Identity: "S+A Eng." <sip:7783720103@10.10.100.10:5060>
SDP IP: 10.10.100.10
SDP Port: 49154

Detail Data:

---- ETH ----
Destination Address: 00-1B-21-02-CB-DC
Source Address: 00-12-01-A1-D2-00
---- 802.1Q Virtual LAN ----
81-00 = 8100h [33024d] Type
{100. ....}= 04h [004d] Priority
{...0 ....}= 00h [000d] CFI
ID:
{.... 0001}= 01h [001d] bits 11-8
{1111 0100}= F4h [244d] bits 7-0
ID = 1F4h [500d]
08-00 = 800h [2048d] Protocol Type: Internet IP (IPv4)
---- IPv4 ----
---- IP header ----
{0100 ....}= 04h [004d] Version: IP Internet Protocol
{.... 0101}= 05h [005d] Header length: in 32 bit units - must be 5 or more
---- Type of Service ----
{100. ....}= 04h [004d] Precedence: Flash override
{...0 ....}= 00h [000d] Delay: Normal
{.... 1...}= 01h [001d] Throughput: High
{.... .0..}= 00h [000d] Reliability: Normal
{.... ..0.}= 00h [000d] ECN Capable Transport(ECT): No
{.... ...0}= 00h [000d] Congestion Experienced(CE): No
03-CF = 3CFh [975d] Total length: in octets, including header length - must not be less than 20
38-45 = 3845h [14405d] Identification
{0... ....}= 00h [000d] reserved: valid
{.0.. ....}= 00h [000d] Don't Fragment (DF bit): May Fragment
{..0. ....}= 00h [000d] More Fragments (MF bit)
Fragment offset:
{...0 0000}= 00h [000d] bits 12-8
{0000 0000}= 00h [000d] bits 7-0
Fragment offset = 00h [000d]: in 64 bits units
{0101 1111}= 5Fh [095d] Time to live: Hops
{0001 0001}= 11h [017d] protocol: UDP (User Datagram)
9B-1B = 9B1Bh [39707d] Header checksum: checksum is correct
source ip address: 10.10.100.10
destination ip address: 10.254.11.36
---- IP datagram ----
---- UDP ----
---- UDP header ----
13-C4 = 13C4h [5060d] source port
13-C4 = 13C4h [5060d] destination port
03-BB = 3BBh [955d] length: octets include header (not less than 8)
00-00 = 00h [000d] checksum: checksum is not present
---- UDP datagram ----
---- SIP ----
INVITE sip:1113214164878151@10.254.11.36 SIP/2.0
Via: SIP/2.0/UDP 10.10.100.10:5060;rport;branch=z9hG4bK759a7e6b2c22bcb61050fc5f8e6d93be
From: "name" <sip:1344@10.254.11.36>;tag=36d16fdc706f3c4e
To: <sip:1113214164878151@10.254.11.36>
Call-ID: be3e816d5b84db55eac9cd62e22a83ff
CSeq: 989685916 INVITE
Contact: "name" <sip:1344@10.10.100.10:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer,100rel
User-Agent: IP Office 8.1 (56)
P-Asserted-Identity: "S+A Eng." <sip:7783720103@10.10.100.10:5060>
Content-Length: 299

v=0
o=UserA 3717427243 3212097672 IN IP4 10.10.100.10
s=Session SDP
c=IN IP4 10.10.100.10
t=0 0
m=audio 49154 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
 
Thanks TLpeter
That did indeed work!!
I find it pretty sad that there are so many very qualified avaya guys on this site, but NONE in Toronto....Another issue that our interconnect couldnt resolve
 
Nice :-)


BAZINGA!

I'm not insane, my mother had me tested!

 
Status
Not open for further replies.

Part and Inventory Search

Sponsor

Back
Top