Oversized Invite packets


#1

Good day all,
Is there any way to “slim down” the invite packets from Grandstream handsets and ATA’s, particularly GXP21xx and HT81x.

We have issues with some of our carriers dropping fragmented UDP packets.

This only seems to be an issue with Grandstream INVITE’s, Yealink and Asterisk invites are much smaller.

I have set “Preferred Vocoder” and “Use First Matching Vocoder” so that the invite only includes 2 codecs but the invites are still 1541 bytes, exceeding the MTU value.

Regards,
Jordan

Frame 700: 95 bytes on wire (760 bits), 95 bytes captured (760 bits)
Ethernet II, Src: Grandstr_ba:b5:7d (00:0b:82:xx:xx:xx), Dst: LannerEl_3a:bf:3d (00:90:0b:xx:xx:xx)
Internet Protocol Version 4, Src: 192.168.1.32, Dst: 168.1.x.x
User Datagram Protocol, Src Port: 29056, Dst Port: 5060
    Source Port: 29056
    Destination Port: 5060
    Length: 1541
    Checksum: 0x834e [unverified]
    [Checksum Status: Unverified]
    [Stream index: 0]
    [Timestamps]
        [Time since first frame: 135.033390000 seconds]
        [Time since previous frame: 0.013138000 seconds]
Session Initiation Protocol (INVITE)
    Request-Line: INVITE sip:041xxxxxxx@realm.pbx.cloud-pbx.com.au SIP/2.0
        Method: INVITE
        Request-URI: sip:041xxxxxxx@realm.pbx.cloud-pbx.com.au
            Request-URI User Part: 041xxxxxxx
            Request-URI Host Part: realm.pbx.cloud-pbx.com.au
        [Resent Packet: False]
    Message Header
        Via: SIP/2.0/UDP 103.4.x.x:46508;branch=z9hG4bK1030062762;rport
            Transport: UDP
            Sent-by Address: 103.4.x.x
            Sent-by port: 46508
            Branch: z9hG4bK1030062762
            RPort: rport
        Route: <sip:portal.cloud-pbx.com.au:5060;lr>
            Route URI: sip:portal.cloud-pbx.com.au:5060;lr
                Route Host Part: portal.cloud-pbx.com.au
                Route Host Port: 5060
                Route URI parameter: lr
        From: "110 Zoe" <sip:user_name@realm.pbx.cloud-pbx.com.au>;tag=1533651551
            SIP Display info: "110 Zoe"
            SIP from address: sip:user_name@realm.pbx.cloud-pbx.com.au
                SIP from address User Part: user_name
                SIP from address Host Part: realm.pbx.cloud-pbx.com.au
            SIP from tag: 1533651551
        To: <sip:041xxxxxxx@realm.pbx.cloud-pbx.com.au>
            SIP to address: sip:041xxxxxxx@realm.pbx.cloud-pbx.com.au
                SIP to address User Part: 041xxxxxxx
                SIP to address Host Part: realm.pbx.cloud-pbx.com.au
        Call-ID: 1692931015-29056-31@BAD.E.CDG.HE
        [Generated Call-ID: 1692931015-29056-31@BAD.E.CDG.HE]
        CSeq: 61 INVITE
            Sequence Number: 61
            Method: INVITE
        Contact: "110 Zoe" <sip:user_name@103.4.x.x:46508>
            SIP Display info: "110 Name"
            Contact URI: sip:user_name@103.4.x.x:46508
                Contact URI User Part: user_name
                Contact URI Host Part: 103.4.x.x
                Contact URI Host Port: 46508
         [truncated]Proxy-Authorization: Digest username="user_name", realm="realm.pbx.cloud-pbx.com.au", nonce="129294ba-a1ed-11e9-b3a2-a5b2b6f54b4a", uri="sip:041xxxxxxx@realm.pbx.cloud-pbx.com.au", response="ad045887f5c9ed1a0d0
            Authentication Scheme: Digest
            Username: "user_name"
            Realm: "realm.pbx.cloud-pbx.com.au"
            Nonce Value: "129294ba-a1ed-11e9-b3a2-a5b2b6f54b4a"
            Authentication URI: "sip:041xxxxxxx@realm.pbx.cloud-pbx.com.au"
            Digest Authentication Response: "ad045887f5c9ed1a0d0636ec8b4d94f7"
            Algorithm: MD5
            CNonce Value: "01255577"
            QOP: auth
            Nonce Count: 00000001
        X-Grandstream-PBX: true
            [Expert Info (Note/Undecoded): Unrecognised SIP header (x-grandstream-pbx)]
                [Unrecognised SIP header (x-grandstream-pbx)]
                [Severity level: Note]
                [Group: Undecoded]
        Max-Forwards: 70
        User-Agent: Grandstream GXP2160 1.0.9.132
        Privacy: none
        P-Preferred-Identity: "110 Name" <sip:user_name@realm.pbx.cloud-pbx.com.au>
            SIP Display info: "110 Name"
            SIP PPI Address: sip:user_name@realm.pbx.cloud-pbx.com.au
                SIP PPI User Part: user_name
                SIP PPI Host Part: realm.pbx.cloud-pbx.com.au
        P-Access-Network-Info: IEEE-EUI-48;eui-48-addr=00-90-0B-xx-xx-xx
            access-type: IEEE-EUI-48
            eui-48-addr=00-90-0B-xx-xx-xx
        P-Emergency-Info: IEEE-EUI-48;eui-48-addr=00-0B-82-xx-xx-xx
            [Expert Info (Note/Undecoded): Unrecognised SIP header (p-emergency-info)]
                [Unrecognised SIP header (p-emergency-info)]
                [Severity level: Note]
                [Group: Undecoded]
        Supported: replaces, path, timer
        Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
        Content-Type: application/sdp
        Accept: application/sdp, application/dtmf-relay
        Content-Length:   247
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): user_mz5q43eunn 8000 8000 IN IP4 103.4.x.x
                Owner Username: user_mz5q43eunn
                Session ID: 8000
                Session Version: 8000
                Owner Network Type: IN
                Owner Address Type: IP4
                Owner Address: 103.4.x.x
            Session Name (s): SIP Call
            Connection Information (c): IN IP4 103.4.x.x
                Connection Network Type: IN
                Connection Address Type: IP4
                Connection Address: 103.4.x.x
            Time Description, active time (t): 0 0
                Session Start Time: 0
                Session Stop Time: 0
            Media Description, name and address (m): audio 15854 RTP/AVP 8 0 101
                Media Type: audio
                Media Port: 15854
                Media Protocol: RTP/AVP
                Media Format: ITU-T G.711 PCMA
                Media Format: ITU-T G.711 PCMU
                Media Format: DynamicRTP-Type-101
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:8 PCMA/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 8
                MIME Type: PCMA
                Sample Rate: 8000
            Media Attribute (a): ptime:20
                Media Attribute Fieldname: ptime
                Media Attribute Value: 20
            Media Attribute (a): rtpmap:0 PCMU/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 0
                MIME Type: PCMU
                Sample Rate: 8000
            Media Attribute (a): rtpmap:101 telephone-event/8000
                Media Attribute Fieldname: rtpmap
                Media Format: 101
                MIME Type: telephone-event
                Sample Rate: 8000
            Media Attribute (a): fmtp:101 0-15
                Media Attribute Fieldname: fmtp
                Media Format: 101 [telephone-event]
                Media format specific parameters: 0-15
            [Generated Call-ID: 1692931015-29056-31@BAD.E.CDG.HE]

#2

The UCM62xx systems have the option to “Send Compact Invite Headers”

Is there anything like this for the HT8xx or GXP21xx devices?